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

[alsa-devel] [PATCH V2] sound: usb: USB3 SuperSpeed sound support
by Paul Zimmerman 14 Aug '10

14 Aug '10

This is V2 of the patch, after feedback from Clemens and Daniel. This patch adds SuperSpeed support to the USB drivers under sound/. It adds tests for USB_SPEED_SUPER to the appropriate places that check for the USB speed. This patch has been tested with our SS USB3 device emulating a set of Yamaha speakers and a Logitech microphone, but with the descriptors modified to add USB3 support. It has also been tested with the real speakers and microphone, to make sure that USB2 devices still work. Signed-off-by: Paul Zimmerman <paulz(a)synopsys.com> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Clemens Ladisch <clemens(a)ladisch.de> Cc: Daniel Mack <daniel(a)caiaq.de> Cc: Greg Kroah-Hartman <gregkh(a)suse.de> --- sound/usb/card.c | 31 +++++++++++++++++++++++-------- sound/usb/helper.c | 17 +++++++++++------ sound/usb/midi.c | 9 ++++++++- sound/usb/pcm.c | 4 ++-- sound/usb/proc.c | 2 +- sound/usb/urb.c | 2 +- 6 files changed, 46 insertions(+), 19 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 7a8ac1d..6c49763 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -299,9 +299,13 @@ static int snd_usb_audio_create(struct usb_device *dev, int idx, *rchip = NULL; - if (snd_usb_get_speed(dev) != USB_SPEED_LOW && - snd_usb_get_speed(dev) != USB_SPEED_FULL && - snd_usb_get_speed(dev) != USB_SPEED_HIGH) { + switch (snd_usb_get_speed(dev)) { + case USB_SPEED_LOW: + case USB_SPEED_FULL: + case USB_SPEED_HIGH: + case USB_SPEED_SUPER: + break; + default: snd_printk(KERN_ERR "unknown device speed %d\n", snd_usb_get_speed(dev)); return -ENXIO; } @@ -377,11 +381,22 @@ static int snd_usb_audio_create(struct usb_device *dev, int idx, if (len < sizeof(card->longname)) usb_make_path(dev, card->longname + len, sizeof(card->longname) - len); - strlcat(card->longname, - snd_usb_get_speed(dev) == USB_SPEED_LOW ? ", low speed" : - snd_usb_get_speed(dev) == USB_SPEED_FULL ? ", full speed" : - ", high speed", - sizeof(card->longname)); + switch (snd_usb_get_speed(dev)) { + case USB_SPEED_LOW: + strlcat(card->longname, ", low speed", sizeof(card->longname)); + break; + case USB_SPEED_FULL: + strlcat(card->longname, ", full speed", sizeof(card->longname)); + break; + case USB_SPEED_HIGH: + strlcat(card->longname, ", high speed", sizeof(card->longname)); + break; + case USB_SPEED_SUPER: + strlcat(card->longname, ", super speed", sizeof(card->longname)); + break; + default: + break; + } snd_usb_audio_create_proc(chip); diff --git a/sound/usb/helper.c b/sound/usb/helper.c index d48d6f8..f280c19 100644 --- a/sound/usb/helper.c +++ b/sound/usb/helper.c @@ -103,11 +103,16 @@ int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe, __u8 request, unsigned char snd_usb_parse_datainterval(struct snd_usb_audio *chip, struct usb_host_interface *alts) { - if (snd_usb_get_speed(chip->dev) == USB_SPEED_HIGH && - get_endpoint(alts, 0)->bInterval >= 1 && - get_endpoint(alts, 0)->bInterval <= 4) - return get_endpoint(alts, 0)->bInterval - 1; - else - return 0; + switch (snd_usb_get_speed(chip->dev)) { + case USB_SPEED_HIGH: + case USB_SPEED_SUPER: + if (get_endpoint(alts, 0)->bInterval >= 1 && + get_endpoint(alts, 0)->bInterval <= 4) + return get_endpoint(alts, 0)->bInterval - 1; + break; + default: + break; + } + return 0; } diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 4678564..6b42ce6 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -834,7 +834,14 @@ static void snd_usbmidi_us122l_output(struct snd_usb_midi_out_endpoint *ep, if (!ep->ports[0].active) return; - count = snd_usb_get_speed(ep->umidi->dev) == USB_SPEED_HIGH ? 1 : 2; + switch (snd_usb_get_speed(ep->umidi->dev)) { + case USB_SPEED_HIGH: + case USB_SPEED_SUPER: + count = 1; + break; + default: + count = 2; + } count = snd_rawmidi_transmit(ep->ports[0].substream, urb->transfer_buffer, count); diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 4568298..ebd09ac 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -467,7 +467,7 @@ static int hw_check_valid_format(struct snd_usb_substream *subs, return 0; } /* check whether the period time is >= the data packet interval */ - if (snd_usb_get_speed(subs->dev) == USB_SPEED_HIGH) { + if (snd_usb_get_speed(subs->dev) != USB_SPEED_FULL) { ptime = 125 * (1 << fp->datainterval); if (ptime > pt->max || (ptime == pt->max && pt->openmax)) { hwc_debug(" > check: ptime %u > max %u\n", ptime, pt->max); @@ -735,7 +735,7 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre } param_period_time_if_needed = SNDRV_PCM_HW_PARAM_PERIOD_TIME; - if (snd_usb_get_speed(subs->dev) != USB_SPEED_HIGH) + if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL) /* full speed devices have fixed data packet interval */ ptmin = 1000; if (ptmin == 1000) diff --git a/sound/usb/proc.c b/sound/usb/proc.c index f5e3f35..3c650ab 100644 --- a/sound/usb/proc.c +++ b/sound/usb/proc.c @@ -107,7 +107,7 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s } snd_iprintf(buffer, "\n"); } - if (snd_usb_get_speed(subs->dev) == USB_SPEED_HIGH) + if (snd_usb_get_speed(subs->dev) != USB_SPEED_FULL) snd_iprintf(buffer, " Data packet interval: %d us\n", 125 * (1 << fp->datainterval)); // snd_iprintf(buffer, " Max Packet Size = %d\n", fp->maxpacksize); diff --git a/sound/usb/urb.c b/sound/usb/urb.c index de607d4..8deeaad 100644 --- a/sound/usb/urb.c +++ b/sound/usb/urb.c @@ -244,7 +244,7 @@ int snd_usb_init_substream_urbs(struct snd_usb_substream *subs, else subs->curpacksize = maxsize; - if (snd_usb_get_speed(subs->dev) == USB_SPEED_HIGH) + if (snd_usb_get_speed(subs->dev) != USB_SPEED_FULL) packs_per_ms = 8 >> subs->datainterval; else packs_per_ms = 1; -- Paul

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Re: [alsa-devel] intel 8x0 ALSA driver in virtualbox - choppy sound
by Takashi Iwai 13 Aug '10

13 Aug '10

At Fri, 13 Aug 2010 17:37:33 +0100, Lex F wrote: > > Hi > > Thanks for your reply. I tried to answer your questions below. > > On 13 August 2010 17:06, Takashi Iwai <tiwai(a)suse.de> wrote: > > We have had many problems with VMware, too :) > > Which sound backend are you using? Direct ALSA / dmix or PA? > > I believe it's direct ALSA, I have choppy output even with > aplay /usr/share/sounds/alsa/*.wav It doesn't mean always that you use direct ALSA. Try to pass -Dplughw option to aplay. > > Actually, there is no real interrupt, but just the virtual one, > > thus it can be never accurate. So, rather the question is how the > > driver is tolerant for sloppy IRQ updates. > > > > Usually, when the period size is large and the buffer have many > > periods (not two), the delayed update works more stably. > > I am afraid I don't understand this :) The comment above is rather to Jaroslav. > I am compiling a kernel with the following options enabled: > CONFIG_SND_PCM_XRUN_DEBUG, CONFIG_SND_VERBOSE_PROCFS, > CONFIG_SND_DEBUG > > Would any of these help to determine the cause and suggest a fix? Fix VirtualBox :) Or try to set xrun_debug and check kernel log as Jaroslav wrote. Takashi

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[alsa-devel] n810 diablo and alsa ?
by John Simpson 13 Aug '10

13 Aug '10

Is there a how-to for installing the alsa-lib correctly on the n810? (hopefully from a fresh install of Maemo/Diablo?) I've been hitting brick walls in every direction. :-( The alsa-lib in the diablo repository seems to be outdated: [1|user@tablet|~/src]cat alsa-lib_1.0.14sa-2maemo2.dsc Format: 1.0 Source: alsa-lib Version: 1.0.14sa-2maemo2 Binary: lib32asound2-dev, lib64asound2, lib32asound2, libasound2-dev, libasound2, libasound2-doc, libasound2-dbg, lib64asound2-dev Maintainer: Debian ALSA Maintainers <pkg-alsa-devel(a)lists.alioth.debian.org> Architecture: any Standards-Version: 3.7.2 Build-Depends: debhelper (>= 4.2.32), quilt (>= 0.40), autotools-dev, doxygen, libc6-dev-powerpc [ppc64], libc6-dev-i386 [amd64], libc6-dev-ppc64 [powerpc], libc6-dev-s390x [s390], libc6-dev-sparc64 [sparc], lib32gcc1 [amd64 ppc64], libc6-dev (>= 2.5) [armel] Vcs-Browser: http://svn.debian.org/wsvn/pkg-alsa/trunk/alsa-lib/ Uploaders: Jordi Mallach <jordi(a)debian.org>, Mikael Magnusson <mikma(a)users.sourceforge.net>, Elimar Riesebieter <riesebie(a)lxtec.de>, Marc-Andre Lureau <marc-andre.lureau(a)nokia.com> Vcs-Svn: svn://svn.debian.org/pkg-alsa/trunk/alsa-lib

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Re: [alsa-devel] intel 8x0 ALSA driver in virtualbox - choppy sound
by Lex F 13 Aug '10

13 Aug '10

Hi Thanks for your reply. I tried to answer your questions below. On 13 August 2010 17:06, Takashi Iwai <tiwai(a)suse.de> wrote: > We have had many problems with VMware, too :) > Which sound backend are you using? Direct ALSA / dmix or PA? I believe it's direct ALSA, I have choppy output even with aplay /usr/share/sounds/alsa/*.wav > Actually, there is no real interrupt, but just the virtual one, > thus it can be never accurate. So, rather the question is how the > driver is tolerant for sloppy IRQ updates. > > Usually, when the period size is large and the buffer have many > periods (not two), the delayed update works more stably. I am afraid I don't understand this :) I am compiling a kernel with the following options enabled: CONFIG_SND_PCM_XRUN_DEBUG, CONFIG_SND_VERBOSE_PROCFS, CONFIG_SND_DEBUG Would any of these help to determine the cause and suggest a fix? Lex

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[alsa-devel] [PATCH 3/5] ASoC: update setting for pxa ssp slave mode
by Haojian Zhuang 13 Aug '10

13 Aug '10

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[alsa-devel] ASoC updates for 2.6.36
by Mark Brown 13 Aug '10

13 Aug '10

The following changes since commit 2e2211a387e70c62e1249822a94f4a33933ecd62: ASoC: add AD1980 obsolete information (2010-08-10 15:43:45 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.36 A couple of focused patches for the WM8994, plus the MAINTAINERS additions. They're not 100% ASoC but mostly so by volume of code covered so hopefully not a problem for you to take via your tree - let me know if it's an issue and I'll drop them. Mark Brown (4): ASoC: Optimise DSP performance for WM8994 ASoC: Only tweak WM8994 chip configuration on devices up to rev D MAINTAINERS: Make Wolfson entry also cover CODEC drivers MAINTAINERS: Add Ian Lartey as comaintaner for Wolfson devices MAINTAINERS | 7 ++++--- sound/soc/codecs/wm8994.c | 23 +++++++++++++++-------- 2 files changed, 19 insertions(+), 11 deletions(-)

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[alsa-devel] [PATCH 5/5] ASoc: add machine driver for 88pm860x
by Haojian Zhuang 13 Aug '10

13 Aug '10

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[alsa-devel] [PATCH 4/5] ASoC: add 88pm860x codec driver
by Haojian Zhuang 13 Aug '10

13 Aug '10

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[alsa-devel] alsa-lib: Rate conversion broken for non-S16?
by Mark Hills 13 Aug '10

13 Aug '10

I have a device (snd-usb-caiaq) whose native sample format is S24_3BE. When sample rate conversion is used, the audio becomes near silent, distorted, and only in the left channel. It is reproducable with aplay (see example below). Recording is not affected. Playback is correct if only format conversion is used. It looks something is causing the incorrect put function to be used by the rate conversion. I traced it to linear_expand() and others, alsa-lib-src/pcm_rate_linear.c. Forcing the function to put S16 format into the destination buffer allows, suprisingly, the sound to be heard correctly and at full volume from both aplay and xwax: if (rate->put_idx == 10) rate->put_idx = 6 I suspected that linear_init() is looking in the wrong place to choose the attributes for the rate conversion. After some time spent I cannot see it. But is there any reason that the output buffer for the rate conversion would not be in S24_3BE? And for the rate conversion to be misinformed? $ uname -s -r -v -p -i -m Linux 2.6.34.1-mh #83 SMP PREEMPT Wed Jul 21 23:25:33 BST 2010 i686 pentium4 i386 $ alsacap Card 0, ID `Audio8DJ', name `Audio 8 DJ' Device 0, ID `Audio 8 DJ', name `Audio 8 DJ', 4 subdevices (4 available) 2 channels, sampling rate 44100..96000 Hz Sample formats: S24_3BE Subdevice 0, name `subdevice #0' Subdevice 1, name `subdevice #1' Subdevice 2, name `subdevice #2' Subdevice 3, name `subdevice #3' $ aplay -v -D plughw:Audio8DJ,0,0 halcyon-24000.wav Playing WAVE 'halcyon-24000.wav' : Signed 16 bit Little Endian, Rate 24000 Hz, Stereo Plug PCM: Rate conversion PCM (48000, sformat=S24_3BE) Converter: linear-interpolation Protocol version: 10002 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 24000 exact rate : 24000 (24000/1) msbits : 16 buffer_size : 10922 period_size : 2730 period_time : 113770 tstamp_mode : NONE period_step : 1 avail_min : 2730 period_event : 0 start_threshold : 10922 stop_threshold : 10922 silence_threshold: 0 silence_size : 0 boundary : 715784192 Slave: Hardware PCM card 0 'Audio 8 DJ' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S24_3BE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 24 buffer_size : 21845 period_size : 5461 period_time : 113770 tstamp_mode : NONE period_step : 1 avail_min : 5461 period_event : 0 start_threshold : 21844 stop_threshold : 21845 silence_threshold: 0 silence_size : 0 boundary : 1431633920 appl_ptr : 0 hw_ptr : 0 -- Mark

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[alsa-devel] [PATCH 0/5] ASoc: support 88pm860x codec driver
by Haojian Zhuang 13 Aug '10

13 Aug '10

Add 88pm860x codec driver. This series patch is based on ASoC for-2.6.37 branch. Thanks Haojian

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