April 2015 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] snd-bebob and PrismSound Orpheus fails with snd_pcm_hw_params_set_channels
by Mike Ood 25 Apr '15

25 Apr '15

Takashi Sakamoto wrote: > > % cat /proc/asound/card2/firewire/clock > > Sampling rate: 44100 > > Clock Source: External (MSU-dest: 7) > > According to this output, the internal clock of your device is set to > synchronize to external source such as S/PDIF or word clock. Oops. Let me boot my Windows7 again, and see.. https://spinorlab.wordpress.com/2015/04/24/orpheus-and-its-sync-source/ (see orpheussyncsource.jpg) The control program by PrismSound tells that the Sync Source is Local, which means it is a Master and System is sync'ed to Orpheus's local clock. http://www.prismsound.com/music_recording/products_subs/orpheus/online_manu… (Contents -> Orpheus hardware -> Synchronization) (Contents -> Orpheus software -> Orpheus Control Panel) Funny? Now reboot again, and going back to Linux. ------------------ % dmesg <- Orpheus was powered on already. ------------------ [ 20.927594] r8169 0000:02:00.0 eth0: link down [ 23.299665] r8169 0000:02:00.0 eth0: link up [ 27.676032] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 29.974533] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 32.282923] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 34.961027] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 37.259624] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 47.732497] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 50.030918] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 52.339515] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 55.037594] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 57.316079] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 ------------------ % ls -l /proc/asound lrwxrwxrwx 1 root root 5 Apr 24 20:24 Orpheus -> card0 ------------------ % cat /proc/asound/card0/firewire/clock Sampling rate: 44100 Clock Source: External (MSU-dest: 7) <- as Takashi pointed out! ------------------ % aplay -D plughw:0,0 test.wav <- same happens as is expected ------------------ Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo aplay: set_params:1297: Unable to install hw params: ACCESS: RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 44100 PERIOD_TIME: (46439 46440) PERIOD_SIZE: 2048 PERIOD_BYTES: 8192 PERIODS: 4 BUFFER_TIME: (185759 185760) BUFFER_SIZE: 8192 BUFFER_BYTES: 32768 TICK_TIME: 0 ------------------ % dmesg <- the last line is added. ------------------ [ 20.927594] r8169 0000:02:00.0 eth0: link down [ 23.299665] r8169 0000:02:00.0 eth0: link up [ 27.676032] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 29.974533] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 32.282923] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 34.961027] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 37.259624] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 47.732497] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 50.030918] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 52.339515] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 55.037594] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 57.316079] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 1483.932836] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 *** > If your device is implemented with no customization of the > setting and you didn't give any external sources to your > device, the device is not synchronized. > (If you give it correctly, please inform it to us.) I am not sure, but I think I gave it correctly to Local Sync Source. > libffado also supports this device and I guess you can > start streaming with jackd and firewire backend Yes, as I have mentioned, I have been always successful with alsa (vanilla without snd-*) + libffado + jackd. > because libffado implements loose > checker of packet discontinuity. I do not understand techinical details, but the sound was perfect even with 192kHz/24bit audio using libffado. So if it's only a #define TIMER_SOMETHING value_anything type of things... > Currently, libffado has a bug to fail to switch clock > source for usual BeBoB driver. > So we have no way to set the source of clock correctly. > I've tried to fix it but get stuck deeply into > ffado's fault and given up. As for me, clock source settings, etc., which is written into its EEPROM when powered off, can be occasionally set via the control program by PrismSound either on Windows/OS X, only when it becomes necessary. So personally I really do not matter, if I understand the situation correctly. > For our information: > http://subversion.ffado.org/wiki/AvcModels/PrismSoundOrpheus I hope some day this level of information becomes usefull for me. Regards, Mike Ood

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Re: [alsa-devel] snd-bebob and PrismSound Orpheus fails with snd_pcm_hw_params_set_channels
by Mike Ood 25 Apr '15

25 Apr '15

Mike Ood wrote: > As for me, clock source settings, etc., > which is written into its EEPROM when powered off, > can be occasionally set via the control program > by PrismSound either on Windows/OS X, > only when it becomes necessary. > I think I was at least partially wrong. The manual says: http://www.prismsound.com/music_recording/products_subs/orpheus/online_manu… (Contents -> Orpheus hardware -> Stand-alone operation): --------------------------------- Stand-alone operation --------------------------------- It is possible to operate Orpheus without a FireWire connection to a host computer. This is done by setting up the unit as required using the Orpheus Control Panel applet whilst it is connected to a host computer via its FireWire interface, then placing the unit in standby by pressing the standby button, before disconnecting the unit from the host and power source. When the unit is re-powered, and detects that no FireWire connection is active, <========== I missed this part! it reloads the settings which were previously stored. As Takashi Sakamoto mentioned: > Currently, libffado has a bug to fail to switch clock > source for usual BeBoB driver. > So we have no way to set the source of clock correctly. > I've tried to fix it but get stuck deeply into > ffado's fault and given up. this could be a problem for me, too. Anyway, the functionality I expect snd-bebob to have is no more than what is offered by juju+libffado+jack. Mike Ood

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[alsa-devel] [PATCH 1/3] ASoC: tas571x: Add DT binding document
by Kevin Cernekee 24 Apr '15

24 Apr '15

Document the bindings for the soon-to-be-added tas571x driver. Signed-off-by: Kevin Cernekee <cernekee(a)chromium.org> --- .../devicetree/bindings/sound/tas571x.txt | 34 ++++++++++++++++++++++ 1 file changed, 34 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas571x.txt diff --git a/Documentation/devicetree/bindings/sound/tas571x.txt b/Documentation/devicetree/bindings/sound/tas571x.txt new file mode 100644 index 000000000000..ac704c1f143d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas571x.txt @@ -0,0 +1,34 @@ +Texas Instruments TAS5711/TAS5717/TAS5719 stereo power amplifiers + +The codec is controlled through an I2C interface. It also has two other +signals that can be wired up to GPIOs: reset (strongly recommended), and +powerdown (optional). + +Required properties: + +- compatible: "ti,tas5711", "ti,tas5717", or "ti,tas5719" +- reg: The I2C address of the device +- #sound-dai-cells: must be equal to 0 + +Optional properties: + +- reset-gpios: GPIO specifier for the TAS571x's active low reset line +- pdn-gpios: GPIO specifier for the TAS571x's active low powerdown line +- clocks: clock phandle for the MCLK input +- clock-names: should be "mclk" +- VDD-supply: regulator phandle for the AVDD/DVDD/HP_VDD supply +- PVDD-supply: regulator phandle for the PVDD supply + +Example: + + tas5717: audio-codec@2a { + compatible = "ti,tas5717"; + reg = <0x2a>; + #sound-dai-cells = <0>; + + reset-gpios = <&gpio5 1 GPIO_ACTIVE_HIGH>; + pdn-gpios = <&gpio5 2 GPIO_ACTIVE_HIGH>; + + clocks = <&clk_core CLK_I2S>; + clock-names = "mclk"; + }; -- 2.2.0.rc0.207.ga3a616c

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[alsa-devel] [PATCH 00/21] HD-audio controller driver split
by Takashi Iwai 24 Apr '15

24 Apr '15

Hi, here is a patch series to split HD-audio controller driver code to snd-hda-core helper module. It's not targeted for 4.1 but rather for 4.2 together with ASoC HD-audio. There is no real changes from the functionality POV. It's mostly code splits and cleanups. The latest patches are found in topic/hda branch of sound git tree. Takashi === Jeeja KP (2): ALSA: hda - moved alloc/free stream pages function to controller library ALSA: hda - add generic functions to set hdac stream params Ramesh Babu (1): ALSA: hda - add ASoC device type for hda core Takashi Iwai (18): ALSA: hda - Handle error from get_response bus ops directly ALSA: hda - Add the controller helper codes to hda-core module ALSA: hda - Add DSP loader to core library code ALSA: hda - Merge codec and controller helpers ALSA: hda - Move send_cmd / get_response to hdac_bus_ops ALSA: hda - Pass bus io_ops directly from the top-level driver ALSA: hda - Migrate hdac_stream into legacy driver ALSA: hda - Migrate more hdac_stream codes ALSA: hda - Embed bus into controller object ALSA: hda - Minor refactoring ALSA: hda - Move PCM format and rate handling code to core library ALSA: hda - Add missing inclusion of <linux/clocksource.h> ALSA: hda - Reenable tracepoints for controller ALSA: hda/tegra - Fix build error and warning ALSA: hda - Drop azx_sd_read*/write*() macros ALSA: hda - Replace hda_bus_ops with static binding ALSA: hda - Move prepared flag into struct hdac_stream ALSA: hda - Replace open codes with snd_hdac_stream_set_params() include/sound/hda_register.h | 152 +++++ include/sound/hdaudio.h | 286 ++++++++- sound/hda/Kconfig | 3 + sound/hda/Makefile | 2 +- sound/hda/hdac_bus.c | 20 +- sound/hda/hdac_controller.c | 507 +++++++++++++++ sound/hda/hdac_device.c | 300 +++++++++ sound/hda/hdac_stream.c | 686 ++++++++++++++++++++ sound/pci/hda/Kconfig | 3 - sound/pci/hda/Makefile | 3 +- sound/pci/hda/hda_codec.c | 421 +------------ sound/pci/hda/hda_codec.h | 80 +-- sound/pci/hda/hda_controller.c | 1330 ++++++++------------------------------- sound/pci/hda/hda_controller.h | 270 +------- sound/pci/hda/hda_intel.c | 234 ++++--- sound/pci/hda/hda_intel.h | 1 + sound/pci/hda/hda_intel_trace.h | 4 +- sound/pci/hda/hda_tegra.c | 76 +-- sound/pci/hda/patch_ca0132.c | 7 +- 19 files changed, 2457 insertions(+), 1928 deletions(-) create mode 100644 include/sound/hda_register.h create mode 100644 sound/hda/hdac_controller.c create mode 100644 sound/hda/hdac_stream.c -- 2.3.5

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Re: [alsa-devel] snd-bebob and PrismSound Orpheus fails with snd_pcm_hw_params_set_channels
by Mike Ood 24 Apr '15

24 Apr '15

Clemens wrote: > Does this device (with this cable) work in Windows or OS X? Definitely. Let me now reset my PC so that it will boot from its Windows7 partion. With Audacity and a vendor supplied control software, my Orpheus plays 44.1kHz 2-ch sound perfectly. Also, >>>> I have been using my PrismSound Orpheus successfully with juju+ffado+jackd as: for more than five years, not with the same PC but with the same cable, during which time I have never experienced a trouble. And before doing clean install of Ubuntu 15.04 (kernel 3.16) a few days ago, I confirmed that: % jackd -R -dfirewire -r44100 -p4096 -n3 & % aplay -D pcm.jack test.wav or % alsaplayer -i text -o jack -d "system:playback_1,system:playback_2" test.wav works all right on Ubuntu 14.04 with kernel 3.13. So I have an impression that this is not a hardware but a software problem, perhaps specific to snd-bebob.. *** > At the moment, a missing packet results in the driver just stopping the > stream. There would be better ways to handle this, but I haven't yet > found the time to implement this. Any suggestions will be appreciated. Thank you again for taking your time. Regards, Mike Ood

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Re: [alsa-devel] snd-bebob and PrismSound Orpheus fails with snd_pcm_hw_params_set_channels
by Mike Ood 24 Apr '15

24 Apr '15

Clemens wrote: >> You should replace "hw" with "plughw" in the device name. So you were referring to: % aplay -L plughw:CARD=Orpheus,DEV=0 Orpheus, Orpheus PCM Hardware device with all software conversions > Any error message in the system log? % tail dmesg?.txt ==> dmesg1.txt <== this is just after booting. [ 21.439026] r8169 0000:02:00.0 eth0: link down [ 23.809013] r8169 0000:02:00.0 eth0: link up ==> dmesg2.txt <== this is just after connecting Orpheus. [ 21.439026] r8169 0000:02:00.0 eth0: link down [ 23.809013] r8169 0000:02:00.0 eth0: link up [ 110.079264] firewire_core 0000:08:00.0: phy config: new root=ffc1, gap_count=5 [ 110.343319] firewire_core 0000:08:00.0: phy config: new root=ffc1, gap_count=5 [ 110.580310] firewire_core 0000:08:00.0: created device fw1: GUID 00119800010001f9, S400 [ 113.660273] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 ==> dmesg3.txt <== this is just after % aplay -D plughw:2,0 test.wav [ 21.439026] r8169 0000:02:00.0 eth0: link down [ 23.809013] r8169 0000:02:00.0 eth0: link up [ 110.079264] firewire_core 0000:08:00.0: phy config: new root=ffc1, gap_count=5 [ 110.343319] firewire_core 0000:08:00.0: phy config: new root=ffc1, gap_count=5 [ 110.580310] firewire_core 0000:08:00.0: created device fw1: GUID 00119800010001f9, S400 [ 113.660273] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 128.010593] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 133.416998] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 136.275142] snd-bebob fw1.0: Detect discontinuity of CIP: 50 70 [ 137.654178] snd-bebob fw1.0: Detect discontinuity of CIP: 70 90 % ls -l dmesg?.txt -rw-rw-r-- 1 user1 user1 72728 Apr 24 08:42 dmesg1.txt -rw-rw-r-- 1 user1 user1 73050 Apr 24 08:42 dmesg2.txt -rw-rw-r-- 1 user1 user1 73318 Apr 24 08:43 dmesg3.txt % egrep 'Apr 24 08:42|Apr 24 08:43' /var/log/syslog Apr 24 08:42:24 host gnome-session[1466]: WARN 2015-04-24 08:42:24 unity.dash.gsettingsscopereader GSettingsScopes.cpp:108 Error fetching protocol metadata for scope: social.scope : Valid key file could not be found in search dirs Apr 24 08:42:44 host kernel: [ 110.079264] firewire_core 0000:08:00.0: phy config: new root=ffc1, gap_count=5 Apr 24 08:42:45 host kernel: [ 110.343319] firewire_core 0000:08:00.0: phy config: new root=ffc1, gap_count=5 Apr 24 08:42:45 host kernel: [ 110.580310] firewire_core 0000:08:00.0: created device fw1: GUID 00119800010001f9, S400 Apr 24 08:42:48 host kernel: [ 113.660273] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 Apr 24 08:43:02 host kernel: [ 128.010593] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 Apr 24 08:43:08 host kernel: [ 133.416998] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 Apr 24 08:43:11 host rtkit-daemon[1163]: Supervising 4 threads of 1 processes of 1 users. Apr 24 08:43:11 host rtkit-daemon[1163]: Successfully made thread 1900 of process 1540 (n/a) owned by '1000' RT at priority 5. Apr 24 08:43:11 host rtkit-daemon[1163]: Supervising 5 threads of 1 processes of 1 users. Apr 24 08:43:11 host kernel: [ 136.275142] snd-bebob fw1.0: Detect discontinuity of CIP: 50 70 Apr 24 08:43:12 host kernel: [ 137.654178] snd-bebob fw1.0: Detect discontinuity of CIP: 70 90 Apr 24 08:43:14 host pulseaudio[1540]: [alsa-source-BeBoB] alsa-util.c: Could not recover from POLLERR|POLLNVAL|POLLHUP and XRUN: Connection timed out Apr 24 08:43:17 host systemd[1]: Stopping User Manager for UID 112... *** Any clues? Mike Ood -- % uname -a Linux host 3.19.0-15-generic #15-Ubuntu SMP Thu Apr 16 23:32:37 UTC 2015 x86_64 x86_64 x86_64 GNU/Linux --

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[alsa-devel] [PATCH] ASoC: rt5677: add register patch for PLL
by Bard Liao 24 Apr '15

24 Apr '15

The PLL output will be unstable in some cases. We can fix it by setting some registers. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5677.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index af18258..9c07a0c 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -62,6 +62,9 @@ static const struct reg_default init_list[] = { {RT5677_PR_BASE + 0x1e, 0x0000}, {RT5677_PR_BASE + 0x12, 0x0eaa}, {RT5677_PR_BASE + 0x14, 0x018a}, + {RT5677_PR_BASE + 0x15, 0x0490}, + {RT5677_PR_BASE + 0x38, 0x0f71}, + {RT5677_PR_BASE + 0x39, 0x0f71}, }; #define RT5677_INIT_REG_LEN ARRAY_SIZE(init_list) -- 1.8.1.1.439.g50a6b54

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Re: [alsa-devel] snd-bebob and PrismSound Orpheus fails with snd_pcm_hw_params_set_channels
by Mike Ood 24 Apr '15

24 Apr '15

I am now trying with my capture program, "myprog.cpp" instead of "aplay.c". ------------------------------- % alsacap ; alscap -R ------------------------------- Card 2, ID `Orpheus', name `Orpheus' Device 0, ID `BeBoB', name `Orpheus PCM', 1 subdevices (1 available) 12 channels, sampling rate 44100..44100 Hz Sample formats: S16_LE, S32_LE Subdevice 0, name `subdevice #0' Card 2, ID `Orpheus', name `Orpheus' Device 0, ID `BeBoB', name `Orpheus PCM', 1 subdevices (1 available) 10 channels, sampling rate 44100..44100 Hz Sample formats: S32_LE Subdevice 0, name `subdevice #0' ------------------------------- % myprog hw:2,0 44100 10 ------------------------------- Sound::Sound: sound_device = hw:2,0, rate = 44100, channels = 10 Sound::asound_set_hwparams: begin... Sound::asound_set_hwparams: buffer_time = 92879, dir = 1 Sound::asound_set_hwparams: buffer_size = 4096 Sound::asound_set_hwparams: period_time = 46439 , dir = 1 Sound::asound_set_hwparams: period_size = 2048, dir = 0 Sound::asound_set_hwparams: Unable to set hw params: Connection timed out Sound::Sound: setting of hwparams failed. Connection timed out ------------------------------- Sound::asound_set_hwparams(snd_pcm_t *, snd_pcm_hw_params_t *) ------------------------------- err = snd_pcm_hw_params_any(handle, params); err = snd_pcm_hw_params_set_rate_resample(handle, params, resample); err = snd_pcm_hw_params_set_access(handle, params, err = snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S32_LE); //if Orphesu err = snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16); //else err = snd_pcm_hw_params_set_channels(handle, params, channels); err = snd_pcm_hw_params_set_rate_near(handle, params, &rrate, 0); err = snd_pcm_hw_params_set_buffer_time_near(handle, params, &buffer_time, err = snd_pcm_hw_params_get_buffer_size(params, &size); err = snd_pcm_hw_params_set_period_time_near(handle, params, &period_time, err = snd_pcm_hw_params_get_period_size(params, &size, &dir); err = snd_pcm_hw_params(handle, params); if (err < 0) { cout << "Sound::asound_set_hwparams: Unable to set hw params: " << snd_strerror(err) << endl; return err; } ------------------------------- No messages were found in % demsg and % cat /var/log/syslog concerning the above run of % myprog. So is it (partly) due to "Connection timed out"? Mike Ood

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[alsa-devel] [PATCH] ALSA: hda - fix headset mic detection problem for one more machine
by Hui Wang 24 Apr '15

24 Apr '15

We have two machines with alc256 codec in the pin quirk table, so moving the common pins to ALC256_STANDARD_PINS. Cc: stable(a)vger.kernel.org BugLink: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1447909 Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 24 +++++++++++++++--------- 1 file changed, 15 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 03975d03..2a72da7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5228,6 +5228,16 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {0x1b, 0x411111f0}, \ {0x1e, 0x411111f0} +#define ALC256_STANDARD_PINS \ + {0x12, 0x90a60140}, \ + {0x14, 0x90170110}, \ + {0x19, 0x411111f0}, \ + {0x1a, 0x411111f0}, \ + {0x1b, 0x411111f0}, \ + {0x1d, 0x40700001}, \ + {0x1e, 0x411111f0}, \ + {0x21, 0x02211020} + #define ALC282_STANDARD_PINS \ {0x14, 0x90170110}, \ {0x18, 0x411111f0}, \ @@ -5328,15 +5338,11 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x1d, 0x40700001}, {0x21, 0x02211050}), SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - {0x12, 0x90a60140}, - {0x13, 0x40000000}, - {0x14, 0x90170110}, - {0x19, 0x411111f0}, - {0x1a, 0x411111f0}, - {0x1b, 0x411111f0}, - {0x1d, 0x40700001}, - {0x1e, 0x411111f0}, - {0x21, 0x02211020}), + ALC256_STANDARD_PINS, + {0x13, 0x40000000}), + SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC256_STANDARD_PINS, + {0x13, 0x411111f0}), SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4, {0x12, 0x90a60130}, {0x13, 0x40000000}, -- 1.9.1

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[alsa-devel] [ANNOUNCE] Audio Mini Summit 2014, Oct. 14, Düsseldorf, Germany
by Takashi Iwai 24 Apr '15

24 Apr '15

Hi, like previous years, we're going to hold a meeting to discuss lowlevel audio on Linux. This time it's along with Linux Plumbers conference, but for avoiding the conflicts with other LPC mini conferences, it'll be held on Tuesday October 14, at the same venue as LPC, Congress Center Düsseldorf, Germany. If anyone is interested, feel free to sign up in the attendee list below. It's a public accessible Google doc, so you can edit freely. http://goo.gl/VbXLPW (or https://docs.google.com/spreadsheets/d/1wBAiAyZSnMjVFJbxdXE5JoVdKGIBIWt-4tu…) The topics that have been currently raised are found in the URL below. If you have any more topics to discuss there, just put your favorite one in the URL below as well. http://goo.gl/JSasQ9 (or https://docs.google.com/spreadsheets/d/1cXzON65QYNrTXi31AIQHffUZ9NWwJuNEJAC…) Thanks! Takashi

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