October 2013 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] [Alsa-user] hdmi sound rerouted?
by Clemens Ladisch 08 Oct '13

08 Oct '13

Dagg Stompler wrote: >> daggs(a)gmx.com wrote: >>> card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] >>> card 1: Generic [HD-Audio Generic], device 3: HDMI 0 [HDMI 0] >>> card 2: HDMI [HDA ATI HDMI], device 3: HDMI 0 [HDMI 0] >> >> The second one is called "Generic" because your kernel does not (yet) >> know the hardware. >> >>> 01:00.1 Audio device: Advanced Micro Devices, Inc. [AMD/ATI] Cedar HDMI Audio [Radeon HD 5400/6300 Series] >>> 06:00.1 Audio device: Advanced Micro Devices, Inc. [AMD/ATI] RV635 HDMI Audio [Radeon HD 3600 Series] >> >> These are the HDMI outputs of *two* graphics cards. The first one's >> model is so new that the driver doesn't know it. > > the 3600 doesn't have hdmi socket so I think it is irrelevant for my issue. The RV635 chip does support HDMI outputs (even if none are actually connected), so the corresponding PCI device shows up here. >>> If you update the kernel, you will get two devices named "HDA ATI HDMI". > > upgrade to where? Oops. Even the latest driver lacks the PCI IDs of many HDMI HDA devices (aa50/aa58/aa60/aa68/aa80/aa88/aa90/aa98: Cypress/Juniper/Redwood/Cedar/ Cayman+Antilles/Barts/Turks/Caicos). Alex, are these devices that require disabling snooping? Regards, Clemens

4 5

[alsa-devel] ALSA: snd-usb-usx2y: remove bogus frame checks
by Guido Aulisi 08 Oct '13

08 Oct '13

Hi, I did some more testing on my US122: I used my patch (see below) and it worked (with default nrpack=4), but I got "should not be here with counts=42" errors, then after some time the kernel crashed, but I couldn't copy the trace. Then I set nrpack=1 and I got no "should not be here..." and it was more stable, no crash. I only checked playback, not recording, nor MIDI, with aplay I played some songs for about one hour... I hope this can be useful. But I think there are other bugs in this driver. Ciao Guido Aulisi The patch is: diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 63fb521..cd16fcc 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -328,7 +328,7 @@ static void i_usX2Y_urb_complete(struct urb *urb) usX2Y_error_urb_status(usX2Y, subs, urb); return; } - if (likely((urb->start_frame & 0xFFFF) == (usX2Y->wait_iso_frame & 0xFFFF))) + if (likely((urb->start_frame & 0x03FF) == (usX2Y->wait_iso_frame & 0x03FF))) subs->completed_urb = urb; else { usX2Y_error_sequence(usX2Y, subs, urb);

2 1

[alsa-devel] Can't get capturing from RME Hammerfall MADI card to work: sound loops & silcence
by Niels Laukens 08 Oct '13

08 Oct '13

Hi, I'm trying to capture audio from an RME Hammerfall MADI card (PCI ID 10ee:3fc6). The setup previously worked fine on Windows, so I'm currently assuming that the card and the input signal is OK. I wrote a very simple ALSA-application that simply opens the device, sets the access format to NONINTERLEAVED (which seems to be the only option the card likes), selects 32bit sampling (although the audio is 24bit, ALSA rejects this option), 48kHz sampling and 64 channels (also, the only option the card accepts). Next, it loops with snd_pcm_readn() to get 100ms worth of audio, and dumps channel 10 to file. (Full source in attachement) The result is strange for various reasons: * The second half of every buffer returned by snd_pcm_readn() is always zero's. * The call seems to return way to fast: I would expect to get 48000 frames/second, but judging from the timing of the snd_pcm_readn() call, it's more like 1536000 frames/second (i.e. 32 fold faster). * When playing back the resulting file, I do recognize some music, but it sounds like it's the same part over and over again. Also, restarting the capture program minutes later, returns the same (at least as far as my ears & brain can recall) fragment of audio. What am I doing wrong here? (Samples of the PCM-file are available upon request) I've also tried a related test with `arecord`, with similar but different results: * The audio was also interspersed with silence * The capture duration did not match the requested '-d' option: the recording was a factor of 2 shorter than requested. * There was more audio in the recording: if I manually cut out the silent parts, the audio sounded reasonable. What am I missing here? Is this the right place to ask? Or should this go to the RME forum? Thanks in advance, Niels

6 5

[alsa-devel] Audio Miniconference 2013 schedule - Edinburgh 21st October
by Mark Brown 07 Oct '13

07 Oct '13

The audio mini-summit in Edinburgh is now less than a month away so it seems like a good time to start pulling together a schedule. Looking at the topics people mentioned when filling out the registration form there were three major topics that people wanted to cover: - Compressed audio review: tinycompress, integration into userspace, driver upstreaming. - UCM and other use case management status & review. - Effects offload to DSPs with dynamically instantiatable algorithms, and general dynamic DSP firmware handling. Plus a few things that only one or two people brought up: - Process stuff (the list, bug tracking and so on). - Some relatively specific hardware support topics: - HDMI hotplug handling throughout the stack - Mic status LED handling throughout the stack. - Code cleanups: - Moving away from the custom ALSA device registration infrastructure. - General code modernisation. - General routing control in userspace - UIs, automation. The first question is if there is anything important that is missing from the above list - please speak up soon if there is! It'd also be good to have volunteers to lead the discussions in each area. In terms of format I'd propose that we go through the list above in roughly that order, having someone lead discussion on the topic. Does this sound good to people? As previously mentioned it'd be good if anyone planning to attend could fill in the form at: https://docs.google.com/forms/d/105uav9tkwwkDU4P1T3IOWjfsa3kgtDbFAJOlHgBQR5… to help with planning & coordination.

8 13

[alsa-devel] [PATCH v2] ALSA: snd-usb-usx2y: remove bogus frame checks
by Daniel Mack 07 Oct '13

07 Oct '13

The frame check in i_usX2Y_urb_complete() and i_usX2Y_usbpcm_urb_complete() is bogus and produces false positives as described in this LAU thread: http://linuxaudio.org/mailarchive/lau/2013/5/20/200177 This patch removes the check code entirely. Cc: fzu(a)wemgehoertderstaat.de Reported-by: Dr Nicholas J Bailey <nicholas.bailey(a)glasgow.ac.uk> Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/usb/usx2y/usbusx2yaudio.c | 22 +++------------------- sound/usb/usx2y/usx2yhwdeppcm.c | 7 +------ 2 files changed, 4 insertions(+), 25 deletions(-) diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 63fb521..6234a51 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -299,19 +299,6 @@ static void usX2Y_error_urb_status(struct usX2Ydev *usX2Y, usX2Y_clients_stop(usX2Y); } -static void usX2Y_error_sequence(struct usX2Ydev *usX2Y, - struct snd_usX2Y_substream *subs, struct urb *urb) -{ - snd_printk(KERN_ERR -"Sequence Error!(hcd_frame=%i ep=%i%s;wait=%i,frame=%i).\n" -"Most probably some urb of usb-frame %i is still missing.\n" -"Cause could be too long delays in usb-hcd interrupt handling.\n", - usb_get_current_frame_number(usX2Y->dev), - subs->endpoint, usb_pipein(urb->pipe) ? "in" : "out", - usX2Y->wait_iso_frame, urb->start_frame, usX2Y->wait_iso_frame); - usX2Y_clients_stop(usX2Y); -} - static void i_usX2Y_urb_complete(struct urb *urb) { struct snd_usX2Y_substream *subs = urb->context; @@ -328,12 +315,9 @@ static void i_usX2Y_urb_complete(struct urb *urb) usX2Y_error_urb_status(usX2Y, subs, urb); return; } - if (likely((urb->start_frame & 0xFFFF) == (usX2Y->wait_iso_frame & 0xFFFF))) - subs->completed_urb = urb; - else { - usX2Y_error_sequence(usX2Y, subs, urb); - return; - } + + subs->completed_urb = urb; + { struct snd_usX2Y_substream *capsubs = usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE], *playbacksubs = usX2Y->subs[SNDRV_PCM_STREAM_PLAYBACK]; diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c index f2a1acd..814d0e8 100644 --- a/sound/usb/usx2y/usx2yhwdeppcm.c +++ b/sound/usb/usx2y/usx2yhwdeppcm.c @@ -244,13 +244,8 @@ static void i_usX2Y_usbpcm_urb_complete(struct urb *urb) usX2Y_error_urb_status(usX2Y, subs, urb); return; } - if (likely((urb->start_frame & 0xFFFF) == (usX2Y->wait_iso_frame & 0xFFFF))) - subs->completed_urb = urb; - else { - usX2Y_error_sequence(usX2Y, subs, urb); - return; - } + subs->completed_urb = urb; capsubs = usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE]; capsubs2 = usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE + 2]; playbacksubs = usX2Y->subs[SNDRV_PCM_STREAM_PLAYBACK]; -- 1.8.3.1

4 10

[alsa-devel] [PATCH] ASoC: tlv320aic3x: Connect 'Left Line1R Mux' and 'Right Line1L Mux'
by Peter Ujfalusi 07 Oct '13

07 Oct '13

The two paths were not connected in the DAPM route causing the associated routes to be non working and the following warnings printed in the logs: tlv320aic3x-codec 0-001b: ASoC: mux Right Line1L Mux has no paths tlv320aic3x-codec 0-001b: ASoC: mux Left Line1R Mux has no paths Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/tlv320aic3x.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 892c108..a160b7d 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -678,6 +678,8 @@ static const struct snd_soc_dapm_route intercon[] = { /* Left Input */ {"Left Line1L Mux", "single-ended", "LINE1L"}, {"Left Line1L Mux", "differential", "LINE1L"}, + {"Left Line1R Mux", "single-ended", "LINE1R"}, + {"Left Line1R Mux", "differential", "LINE1R"}, {"Left Line2L Mux", "single-ended", "LINE2L"}, {"Left Line2L Mux", "differential", "LINE2L"}, @@ -694,6 +696,8 @@ static const struct snd_soc_dapm_route intercon[] = { /* Right Input */ {"Right Line1R Mux", "single-ended", "LINE1R"}, {"Right Line1R Mux", "differential", "LINE1R"}, + {"Right Line1L Mux", "single-ended", "LINE1L"}, + {"Right Line1L Mux", "differential", "LINE1L"}, {"Right Line2R Mux", "single-ended", "LINE2R"}, {"Right Line2R Mux", "differential", "LINE2R"}, -- 1.8.3.2

2 1

Re: [alsa-devel] [PATCH] sound: usb: support wireless devices in snd_usb_parse_datainterval
by Takashi Iwai 07 Oct '13

07 Oct '13

At Tue, 1 Oct 2013 14:32:57 -0500, Thomas Pugliese wrote: > > Hi, > This patch adds support for dev speed USB_SPEED_WIRELESS in > snd_usb_parse_datainterval which allows the usb sound core to create > ISO urbs with the correct number and size of buffers. > > Signed-off-by: Thomas Pugliese <thomas.pugliese(a)gmail.com> Thanks, applied this one, too. Takashi > --- > sound/usb/helper.c | 1 + > 1 file changed, 1 insertion(+) > > diff --git a/sound/usb/helper.c b/sound/usb/helper.c > index 6209024..51ed1ac 100644 > --- a/sound/usb/helper.c > +++ b/sound/usb/helper.c > @@ -118,6 +118,7 @@ unsigned char snd_usb_parse_datainterval(struct snd_usb_audio *chip, > { > switch (snd_usb_get_speed(chip->dev)) { > case USB_SPEED_HIGH: > + case USB_SPEED_WIRELESS: > case USB_SPEED_SUPER: > if (get_endpoint(alts, 0)->bInterval >= 1 && > get_endpoint(alts, 0)->bInterval <= 4) > -- > 1.7.10.4 > > -- > To unsubscribe from this list: send the line "unsubscribe linux-usb" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html >

1 0

Re: [alsa-devel] [PATCH] sound: usb: add support for wireless USB devices
by Takashi Iwai 07 Oct '13

07 Oct '13

At Tue, 1 Oct 2013 14:32:15 -0500, Thomas Pugliese wrote: > > This patch updates snd_usb_audio_create also support devices whose > speed == USB_SPEED_WIRELESS. > > Signed-off-by: Thomas Pugliese <thomas.pugliese(a)gmail.com> Thanks, applied now. (BTW, please add the maintainers explicitly to Cc at the next time.) Takashi > --- > sound/usb/card.c | 1 + > 1 file changed, 1 insertion(+) > > diff --git a/sound/usb/card.c b/sound/usb/card.c > index 64952e2..ef76aa0 100644 > --- a/sound/usb/card.c > +++ b/sound/usb/card.c > @@ -349,6 +349,7 @@ static int snd_usb_audio_create(struct usb_device *dev, int idx, > case USB_SPEED_LOW: > case USB_SPEED_FULL: > case USB_SPEED_HIGH: > + case USB_SPEED_WIRELESS: > case USB_SPEED_SUPER: > break; > default: > -- > 1.7.10.4 > > -- > To unsubscribe from this list: send the line "unsubscribe linux-usb" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html >

1 0

[alsa-devel] [PATCH 1/3] ASoC: dapm: Add support for virtual mixer controls
by Lars-Peter Clausen 07 Oct '13

07 Oct '13

This patch adds support for virtual DAPM mixer controls. They are similar to virtual DAPM enums. There is no hardware register backing the control, so changing the control's value wont have any direct effect on the hardware. But it still influences the DAPM graph by causing the path it sits on to be connected or disconnected. This in turn can cause power changes for some of the widgets on the DAPM graph, which will then modify the hardware state. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc-dapm.h | 4 ++++ include/sound/soc.h | 3 ++- sound/soc/soc-dapm.c | 45 +++++++++++++++++++++++++++------------------ 3 files changed, 33 insertions(+), 19 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 27a72d5..2037c45 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -286,6 +286,8 @@ struct device; .info = snd_soc_info_volsw, \ .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 1) } +#define SOC_DAPM_SINGLE_VIRT(xname, max) \ + SOC_DAPM_SINGLE(xname, SND_SOC_NOPM, 0, max, 0) #define SOC_DAPM_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, \ @@ -300,6 +302,8 @@ struct device; .tlv.p = (tlv_array), \ .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 0) } +#define SOC_DAPM_SINGLE_TLV_VIRT(xname, max, tlv_array) \ + SOC_DAPM_SINGLE(xname, SND_SOC_NOPM, 0, max, 0, tlv_array) #define SOC_DAPM_ENUM(xname, xenum) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_enum_double, \ diff --git a/include/sound/soc.h b/include/sound/soc.h index 6dd329a..af1678c 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1049,7 +1049,8 @@ struct snd_soc_pcm_runtime { /* mixer control */ struct soc_mixer_control { int min, max, platform_max; - unsigned int reg, rreg, shift, rshift; + int reg, rreg; + unsigned int shift, rshift; unsigned int invert:1; unsigned int autodisable:1; }; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c17c14c..25b9554 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -499,18 +499,22 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, int val; struct soc_mixer_control *mc = (struct soc_mixer_control *) w->kcontrol_news[i].private_value; - unsigned int reg = mc->reg; + int reg = mc->reg; unsigned int shift = mc->shift; int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; - val = soc_widget_read(w, reg); - val = (val >> shift) & mask; - if (invert) - val = max - val; + if (reg != SND_SOC_NOPM) { + val = soc_widget_read(w, reg); + val = (val >> shift) & mask; + if (invert) + val = max - val; + p->connect = !!val; + } else { + p->connect = 0; + } - p->connect = !!val; } break; case snd_soc_dapm_mux: { @@ -2791,7 +2795,7 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, struct snd_soc_card *card = codec->card; struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; + int reg = mc->reg; unsigned int shift = mc->shift; int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; @@ -2804,7 +2808,7 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, kcontrol->id.name); mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - if (dapm_kcontrol_is_powered(kcontrol)) + if (dapm_kcontrol_is_powered(kcontrol) && reg != SND_SOC_NOPM) val = (snd_soc_read(codec, reg) >> shift) & mask; else val = dapm_kcontrol_get_value(kcontrol); @@ -2835,7 +2839,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, struct snd_soc_card *card = codec->card; struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; + int reg = mc->reg; unsigned int shift = mc->shift; int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; @@ -2857,19 +2861,24 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - dapm_kcontrol_set_value(kcontrol, val); + change = dapm_kcontrol_set_value(kcontrol, val); - mask = mask << shift; - val = val << shift; + if (reg != SND_SOC_NOPM) { + mask = mask << shift; + val = val << shift; + + change = snd_soc_test_bits(codec, reg, mask, val); + } - change = snd_soc_test_bits(codec, reg, mask, val); if (change) { - update.kcontrol = kcontrol; - update.reg = reg; - update.mask = mask; - update.val = val; + if (reg != SND_SOC_NOPM) { + update.kcontrol = kcontrol; + update.reg = reg; + update.mask = mask; + update.val = val; - card->update = &update; + card->update = &update; + } soc_dapm_mixer_update_power(card, kcontrol, connect); -- 1.8.0

3 10

[alsa-devel] [PATCH 0/7] ALSA: hda - hdmi: Various channel mapping fixes
by Anssi Hannula 07 Oct '13

07 Oct '13

Hi! Apparently there were several more bugs in the HDMI channel mapping code, so here you go. I haven't marked any for stable since these don't really affect the regular layouts that much and the patches aren't that trivial. If you want to pick some for stable, my suggestions are 1, 5, 6, maybe 3 (those I initially marked for stable until I changed my mind). Anssi Hannula (7): ALSA: hda - hdmi: Fix reported channel map on common default layouts ALSA: hda - hdmi: Fix incorrect default channel mapping for unusual CAs ALSA: hda - hdmi: Fix programmed active channel count ALSA: hda - hdmi: Fix unused slots being enabled in manual and non-PCM mappings ALSA: hda - hdmi: Fix channel maps with less common speakers ALSA: hda - hdmi: Fix available channel maps missing from TLV ALSA: hda - hdmi: Tweak debug messages to be more useful sound/pci/hda/patch_hdmi.c | 146 +++++++++++++++++++++-------------- 1 file changed, 90 insertions(+), 56 deletions(-) -- Anssi Hannula

2 8