May 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] hdspmixer/ubuntu
by Tim Blechmann 12 May '11

12 May '11

hi all, the current version of ubuntu ships alsa-tools without hdspmixer [1], probably because of the problems of older versions with newer kernels (iirc a hardcoded preprocessor definition made the old version unusable). while it is still possible to compile a working hdspmixer from sources, this is quite inconvenient (especially for newbies). questions: - is there a plan to release a 1.0.24.2, including the hdspmixer fixes? - gitcompile currently does not work on ubuntu, but requires this patch [2] in order to run the autotools. according to the name, it is already required for older versions - compiling ld10k1 fails with: make[1]: Entering directory `/home/tim/workspace/alsa-tools/ld10k1' Making all in setup make[2]: Entering directory `/home/tim/workspace/alsa-tools/ld10k1/setup' Making all in effects make[3]: Entering directory `/home/tim/workspace/alsa- tools/ld10k1/setup/effects' make[3]: *** No rule to make target `simple.emu10k1', needed by `all-am'. Stop. is this a problem on my side? thanks, tim [1] https://bugs.launchpad.net/ubuntu/+source/alsa-tools/+bug/723276 [2] http://svnweb.mageia.org/packages/cauldron/alsa-tools/current/SOURCES/alsa- tools-1.0.11-gtk-buildfix.patch?view=log&pathrev=1611 -- tim(a)klingt.org http://tim.klingt.org A: Because it messes up the order in which people normally read text. Q: Why is top-posting such a bad thing? A: Top-posting. Q: What is the most annoying thing in e-mail?

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[alsa-devel] [PATCH] ASoC: Reintroduce do_spi_write()
by Mark Brown 12 May '11

12 May '11

There is an unfortunate difference in return values between spi_write() and i2c_master_send() so we need an adaptor function to translate. Reported-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-cache.c | 16 +++++++++++++++- 1 files changed, 15 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 04861c4..dee86fc 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -20,6 +20,20 @@ #include <trace/events/asoc.h> +#ifdef CONFIG_SPI_MASTER +static int do_spi_write(void *control, const char *data, int len) +{ + struct spi_device *spi = control; + int ret; + + ret = spi_write(spi, data, len); + if (ret < 0) + return ret; + + return len; +} +#endif + static int do_hw_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value, const void *data, int len) { @@ -409,7 +423,7 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, case SND_SOC_SPI: #ifdef CONFIG_SPI_MASTER - codec->hw_write = (hw_write_t)spi_write; + codec->hw_write = do_spi_write; #endif codec->control_data = container_of(codec->dev, -- 1.7.5.1

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[alsa-devel] [PATCH] Enable E-mu 0204
by Joseph Teichman 12 May '11

12 May '11

This is a trivial patch to enable the E-mu 0204, which is the newest device in the series 0202USB, 0404USB and Tracker Pre. Signed-off-by: Joseph Teichman <josteich at gmail.com> --- alsa-kmirror/usb/mixer.c 2011-01-27 07:09:20.000000000 -0500 +++ alsa-kmirror-jt/usb/mixer.c 2011-02-05 20:42:00.000000000 -0500 @@ -1566,7 +1566,7 @@ cval->initialized = 1; } else { if (type == USB_XU_CLOCK_RATE) { - /* E-Mu USB 0404/0202/TrackerPre + /* E-Mu USB 0404/0202/TrackerPre/0204 * samplerate control quirk */ cval->min = 0; diff -u ./quirks.c /home/joe/alsa-driver-1.0.24-jt/sound/usb/quirks.c --- ./quirks.c 2011-01-27 07:09:20.000000000 -0500 +++ /home/joe/alsa-driver-1.0.24-jt/sound/usb/quirks.c 2011-02-05 20:40:28.000000000 -0500 @@ -532,7 +532,7 @@ } /* - * For E-Mu 0404USB/0202USB/TrackerPre sample rate should be set for device, + * For E-Mu 0404USB/0202USB/TrackerPre/0204USB sample rate should be set for device, * not for interface. */ @@ -589,6 +589,7 @@ case USB_ID(0x041e, 0x3f02): /* E-Mu 0202 USB */ case USB_ID(0x041e, 0x3f04): /* E-Mu 0404 USB */ case USB_ID(0x041e, 0x3f0a): /* E-Mu Tracker Pre */ + case USB_ID(0x041e, 0x3f19): /* E-Mu 0204 USB */ set_format_emu_quirk(subs, fmt); break; } diff -u ./quirks-table.h /home/joe/alsa-driver-1.0.24-jt/sound/usb/quirks-table.h --- ./quirks-table.h 2011-01-27 07:09:20.000000000 -0500 +++ /home/joe/alsa-driver-1.0.24-jt/sound/usb/quirks-table.h 2011-02-05 20:41:02.000000000 -0500 @@ -79,6 +79,13 @@ .idProduct = 0x3f0a, .bInterfaceClass = USB_CLASS_AUDIO, }, +{ + /* E-Mu 0204 USB */ + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = 0x041e, + .idProduct = 0x3f19, + .bInterfaceClass = USB_CLASS_AUDIO, +}, /* * Logitech QuickCam: bDeviceClass is vendor-specific, so generic interface

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[alsa-devel] Has hdspmixer been dropped from alsa-tools?
by Lars Zobbe 12 May '11

12 May '11

Hi. Just upgraded to ubuntu 11.04 and went through the usual process with getting HDSP 9632 to work. Everything seems to be fine, modules loaded and stuff. But I can't find the hdspmixer. Normally it's included in the alsa-tools and alsa-tools-gui packages. And actually the mixer GUI does get an entry in the menu. But there is no command for it to execute. It is supposed to run hdspmixer - but it's not there. If I write "hdsp" and try to autocomplete with tab in the terminal, I can now only se hdsploader and hdspconf (both seem to run just fine btw.). Without the alsamixer I won't get any sound. Is the error in the ubuntu natty packaging or did hdspmixer get lost? Is there any way I can install the hdspmixer tool? Would it break anything if I just snatched an old version fx from ubuntu 10.04 repositories (where it IS present)? Some info: 00-00: RME Hammerfall HDSP 9632 : RME Hammerfall HDSP 9632 : playback 1 : capt _From *lspci* 02:09.0 Multimedia audio controller: Xilinx Corporation RME Hammerfall DSP (rev 9a) _From *lshw*: description: Multimedia audio controller product: RME Hammerfall DSP vendor: Xilinx Corporation physical id: 9 bus info: pci@0000:02:09.0 version: 9a width: 32 bits clock: 33MHz capabilities: bus_master configuration: driver=RME Hammerfall DSP latency=255 resources: irq:19 memory:fdff0000-fdffffff *uname -a:* Linux lars-desktop 2.6.38-8-generic #42-Ubuntu SMP Mon Apr 11 03:31:50 UTC 2011 i686 athlon i386 GNU/Linux *Version of alsa-tools*: *** 1.0.24.1-0ubuntu1 0 *Version of alsa-firmware-loaders*: *** 1.0.24.1-0ubuntu1 0 *cat /proc/asound/version * Advanced Linux Sound Architecture Driver Version 1.0.23. *cat /proc/asound/cards* 0 [DSP ]: H-DSP - Hammerfall DSP RME Hammerfall HDSP 9632 at 0xfdff0000, irq 19 *cat /proc/asound/modules * 0 snd_hdsp *cat /proc/asound/DSP/hdsp * RME Hammerfall HDSP 9632 (Card #1) Buffers: capture f2400000 playback f2600000 IRQ: 19 Registers bus: 0xfdff0000 VM: 0xf819c000 Control register: 0x300409a Control2 register: 0x0 Status register: 0x32000000 Status2 register: 0xffff8701 FIFO status: 0 MIDI1 Output status: 0xffffff00 MIDI1 Input status: 0xffffff00 MIDI2 Output status: 0xffffff00 MIDI2 Input status: 0xffffff00 Use Midi Tasklet: on Buffer Size (Latency): 2048 samples (2 periods of 8192 bytes) Hardware pointer (frames): 0 Precise pointer: off Line out: on Firmware version: 1 Sample Clock Source: Internal 44.1 kHz Preferred Sync Reference: ADAT1 AutoSync Reference: None AutoSync Frequency: 0 System Clock Mode: Master System Clock Frequency: 44100 System Clock Locked: No IEC958 input: Coaxial IEC958 output: Coaxial only IEC958 quality: Consumer IEC958 emphasis: off IEC958 NonAudio: off IEC958 sample rate: Error flag set ADAT1: No Lock SPDIF: No Lock Word Clock: No Lock ADAT Sync: No Lock AD Gain : +4 dBu DA Gain : -10 dBV Phones Gain : -12 dB XLR Breakout Cable : no AEB : off (ADAT1 external) *modinfo soundcore* filename: /lib/modules/2.6.38-8-generic/kernel/sound/soundcore.ko license: GPL author: Alan Cox description: Core sound module srcversion: 982D4ED4F640D0B9C43B130 depends: vermagic: 2.6.38-8-generic SMP mod_unload modversions 686 *lsmod | grep snd* snd_hdsp 53247 0 snd_pcm 80244 1 snd_hdsp snd_page_alloc 14073 2 snd_hdsp,snd_pcm snd_hwdep 13274 1 snd_hdsp snd_seq_midi 13132 0 snd_rawmidi 25269 2 snd_hdsp,snd_seq_midi snd_seq_midi_event 14475 1 snd_seq_midi snd_seq 51291 2 snd_seq_midi,snd_seq_midi_event snd_timer 28659 2 snd_pcm,snd_seq snd_seq_device 14110 3 snd_seq_midi,snd_rawmidi,snd_seq snd 55295 7 snd_hdsp,snd_pcm,snd_hwdep,snd_rawmidi,snd_seq,snd_timer,snd_seq_device soundcore 12600 1 snd Sincerly Lars Zobbe.

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[alsa-devel] [PATCH 0/3] ASoC: codecs: max98088: bug fixes and changes
by jinyoungp＠nvidia.com 12 May '11

12 May '11

From: Jin Park <jinyoungp(a)nvidia.com> Recently, I working on max98088 and changed something in max98088 asoc driver. The MAX98088 driver had some bug and needed modifies. The changes was confirmed by Peter Hsiang who author of max98088 driver and maxim engineer. Please review my changes. Thanks, Jin Park. Jin Park (3): ASoC: codecs: max98088: Fixed invalid register definitions in mixer controls ASoC: codecs: max98088: Moved the EX Limiter Mode from dapm widget to control ASoC: codecs: max98088: Added digital mute function in DAI1 and DAI2 sound/soc/codecs/max98088.c | 62 +++++++++++++++++++++++++++++++------------ sound/soc/codecs/max98088.h | 13 +++++++++ 2 files changed, 58 insertions(+), 17 deletions(-) ----------------------------------------------------------------------------------- This email message is for the sole use of the intended recipient(s) and may contain confidential information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -----------------------------------------------------------------------------------

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[alsa-devel] [PATCH 1/2] ASoC: Remove byte swap in 4x12 SPI write
by Mark Brown 11 May '11

11 May '11

snd_soc_4_12_spi_write() contains a byte swap. Since this code was written for an Analog CODEC on a Blackfin reference board it appears that this is done because while Blackfin is little endian the CODEC is big endian (as are most CODECs). Push this up into the generic 4x12 write function and use cpu_to_be16() to do the byte swap so things are more regular and things work on both CPU endiannesses. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-cache.c | 11 +++++------ 1 files changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 6e5e30a..73907e5 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -101,12 +101,11 @@ static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec, static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { - u8 data[2]; + u16 data; - data[0] = (reg << 4) | ((value >> 8) & 0x000f); - data[1] = value & 0x00ff; + data = cpu_to_be16((reg << 12) | (value & 0xffffff)); - return do_hw_write(codec, reg, value, data, 2); + return do_hw_write(codec, reg, value, &data, 2); } #if defined(CONFIG_SPI_MASTER) @@ -115,8 +114,8 @@ static int snd_soc_4_12_spi_write(void *control_data, const char *data, { u8 msg[2]; - msg[0] = data[1]; - msg[1] = data[0]; + msg[0] = data[0]; + msg[1] = data[1]; return do_spi_write(control_data, msg, len); } -- 1.7.5.1

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[alsa-devel] how to link multi plugin to dsnoop ?
by Vincent Meserette 11 May '11

11 May '11

Hi all, I would like to allow multiple applications to access the audio inputs and outputs of my system at same time. I succeded to make it for the audio outputs by using the dmix plugin, but I don't know how to do it with the dsnoop plugin because it seems I cannot link directly the multi plugin to dsnoop. Here is the error message I have with the /etc/asound.conf file presented below : arecord -D"alduplex" -c4 -r48000 -fS16_LE /tmp/test.wav ALSA lib pcm_direct.c:1490:(_snd_pcm_direct_get_slave_ipc_offset) Invalid type 'multi' for slave PCM arecord: main:654: audio open error: Invalid argument Does someone know how can I route the multi plugin to dsnoop ? Thanks in advance My /etc/asound.conf file : #output device pcm.AD1989A_outputs{ type hw card 0 } pcm.aldmix{ type dmix ipc_key 1024 ipc_perm 0666 slave{ pcm AD1989A_outputs } } #input device pcm.4channelsDeinterleaved { type multi slaves.a.pcm "hw:0,0,0" slaves.a.channels 2 slaves.b.pcm "hw:0,0,2" slaves.b.channels 2 bindings.0.slave a bindings.0.channel 0 bindings.1.slave a bindings.1.channel 1 bindings.2.slave b bindings.2.channel 0 bindings.3.slave b bindings.3.channel 1 } pcm.AD1989A_inputs { type route slave.pcm "4channelsDeinterleaved" ttable.0.0 1 ttable.1.1 1 ttable.2.2 1 ttable.3.3 1 } #4 channels dsnoop (plug-in for capture) pcm.aldsnoop{ type dsnoop ipc_key 1025 ipc_perm 0666 slave{ pcm AD1989A_inputs rate 48000 channels 4 } } pcm.alduplex { type asym capture.pcm "aldsnoop" playback.pcm "aldmix" } ### CTL DEVICES (controls) ctl.mixer0 { type hw card 0 }

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[alsa-devel] ASoC update for 2.6.39
by Mark Brown 11 May '11

11 May '11

The following changes since commit 1209842af4db98ffd7364ec9cf7d1a59293aa74c: Merge branch 'for-2.6.40' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc (2011-05-10 09:24:50 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.39 Stephen Warren (1): ASoC: WM8903: Fix Digital Capture Volume range sound/soc/codecs/wm8903.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-)

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[alsa-devel] ASoC update for 2.6.40
by Mark Brown 11 May '11

11 May '11

The following changes since commit 1209842af4db98ffd7364ec9cf7d1a59293aa74c: Merge branch 'for-2.6.40' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc (2011-05-10 09:24:50 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.40 Mark Brown (1): Merge branch 'for-2.6.39' into for-2.6.40 Stephen Warren (1): ASoC: WM8903: Fix Digital Capture Volume range sound/soc/codecs/wm8903.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-)

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[alsa-devel] [PATCH] ASoC: Disable WM8994/58 microphone detection over suspend
by Mark Brown 11 May '11

11 May '11

It will be non-functional with the basises and clocks off anyway, if the system needs microphone detection enabled over suspend then it should be causing the CODEC to ignore suspend using the APIs for that to prevent the biases being disabled. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 25 +++++++++++++++++++++++++ 1 files changed, 25 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index b6d47e7..e6dfa10 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2416,8 +2416,19 @@ static struct snd_soc_dai_driver wm8994_dai[] = { static int wm8994_suspend(struct snd_soc_codec *codec, pm_message_t state) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + struct wm8994 *control = codec->control_data; int i, ret; + switch (control->type) { + case WM8994: + snd_soc_update_bits(codec, WM8994_MICBIAS, WM8994_MICD_ENA, 0); + break; + case WM8958: + snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, + WM8958_MICD_ENA, 0); + break; + } + for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) { memcpy(&wm8994->fll_suspend[i], &wm8994->fll[i], sizeof(struct wm8994_fll_config)); @@ -2435,6 +2446,7 @@ static int wm8994_suspend(struct snd_soc_codec *codec, pm_message_t state) static int wm8994_resume(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + struct wm8994 *control = codec->control_data; int i, ret; unsigned int val, mask; @@ -2473,6 +2485,19 @@ static int wm8994_resume(struct snd_soc_codec *codec) i + 1, ret); } + switch (control->type) { + case WM8994: + if (wm8994->micdet[0].jack || wm8994->micdet[1].jack) + snd_soc_update_bits(codec, WM8994_MICBIAS, + WM8994_MICD_ENA, WM8994_MICD_ENA); + break; + case WM8958: + if (wm8994->jack_cb) + snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, + WM8958_MICD_ENA, WM8958_MICD_ENA); + break; + } + return 0; } #else -- 1.7.5.1

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