February 2012 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] [PATCH 1/6] hda_intel: Add Lynx Point HD Audio Controller DeviceIDs
by Takashi Iwai 08 Feb '12

08 Feb '12

At Wed, 8 Feb 2012 00:35:33 +0000, Heasley, Seth wrote: > > >> I'm also in favor of not rocking the boat, but I'm curious what > >changing this would entail, and what would be the advantage or rationale > >for making the change? > > > >Reading the LPIB register, and reading the DMA position buffer, are two > >different methods of reading the current playback (or recording) > >position, i e which sample is currently being played back. > > > >The problem is that some chipsets prefer one method over the other, i e, > >only one of the methods work reliably. And if the other method only > >breaks occasionally, this can be quite difficult to detect and track > >down - we will have users complaining about their audio sometimes either > >sounding distorted, maybe not working at all, or just once in a while > >glitches. All of these symptoms can have many causes, so deducing that > >to a broken playback position is time consuming. That's why I think it's > >worth the extra question, to get it right from the start. > > > >So...thanks in advance for looking it up for us? :-) > > I'm inquiring about it, but I can say that it seems to work just fine using the DMA position buffer. OK, I merged the patch now as is. Please let me know if you find a problem. thanks, Takashi

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[alsa-devel] [PATCH] Add support for Echo3G devices. Thanks to unimatrix.
by Luke Yelavich 08 Feb '12

08 Feb '12

From: Luke Yelavich <luke.yelavich(a)canonical.com> BugLink: https://bugs.launchpad.net/bugs/433573 Signed-off-by: Luke Yelavich <luke.yelavich(a)canonical.com> --- src/conf/cards/Echo_Echo3G.conf | 318 +++++++++++++++++++++++++++++++++++++++ src/conf/cards/Makefile.am | 1 + 2 files changed, 319 insertions(+), 0 deletions(-) create mode 100644 src/conf/cards/Echo_Echo3G.conf diff --git a/src/conf/cards/Echo_Echo3G.conf b/src/conf/cards/Echo_Echo3G.conf new file mode 100644 index 0000000..766f13f --- /dev/null +++ b/src/conf/cards/Echo_Echo3G.conf @@ -0,0 +1,318 @@ +# +# Configuration for the Echo3G driver +# + +<confdir:pcm/front.conf> +Echo_Echo3G.pcm.front.0 { + @args [ CARD ] + @args.CARD { + type string + } + type hw + card $CARD + device 0 + subdevice 0 +} + +<confdir:pcm/rear.conf> +Echo_Echo3G.pcm.rear.0 { + @args [ CARD ] + @args.CARD { + type string + } + type hw + card $CARD + device 0 + subdevice 4 +} + +<confdir:pcm/center_lfe.conf> +Echo_Echo3G.pcm.center_lfe.0 { + @args [ CARD ] + @args.CARD { + type string + } + type hw + card $CARD + device 0 + subdevice 2 +} + +<confdir:pcm/side.conf> +Echo_Echo3G.pcm.side.0 { + @args [ CARD ] + @args.CARD { + type string + } + type hw + card $CARD + device 0 + subdevice 6 +} + +<confdir:pcm/surround40.conf> +Echo_Echo3G.pcm.surround40.0 { + @args [ CARD ] + @args.CARD { + type string + } + type multi + slaves [ + { + pcm { + @func concat + strings [ + "cards.Echo_Echo3G.pcm.front.0:CARD=" $CARD + ] + } + channels 2 + } + { + pcm { + @func concat + strings [ + "cards.Echo_Echo3G.pcm.rear.0:CARD=" $CARD + ] + } + channels 2 + } + ] + bindings [ + { slave 0 channel 0 } + { slave 0 channel 1 } + { slave 1 channel 0 } + { slave 1 channel 1 } + ] +} + +<confdir:pcm/surround41.conf> +Echo_Echo3G.pcm.surround41.0 { + @args [ CARD ] + @args.CARD { + type string + } + type multi + slaves [ + { + pcm { + @func concat + strings [ + "cards.Echo_Echo3G.pcm.front.0:CARD=" $CARD + ] + } + channels 2 + } + { + pcm { + @func concat + strings [ + "cards.Echo_Echo3G.pcm.rear.0:CARD=" $CARD + ] + } + channels 2 + } + { + pcm { + @func concat + strings [ + "cards.Echo_Echo3G.pcm.center_lfe.0:CARD=" $CARD + ] + } + channels 2 + } + ] + bindings [ + { slave 0 channel 0 } + { slave 0 channel 1 } + { slave 1 channel 0 } + { slave 1 channel 1 } + { slave 2 channel 0 } + ] +} + +<confdir:pcm/surround50.conf> +Echo_Echo3G.pcm.surround50.0 { + @args [ CARD ] + @args.CARD { + type string + } + type multi + slaves [ + { + pcm { + @func concat + strings [ + "cards.Echo_Echo3G.pcm.front.0:CARD=" $CARD + ] + } + channels 2 + } + { + pcm { + @func concat + strings [ + "cards.Echo_Echo3G.pcm.rear.0:CARD=" $CARD + ] + } + channels 2 + } + { + pcm { + @func concat + strings [ + "cards.Echo_Echo3G.pcm.center_lfe.0:CARD=" $CARD + ] + } + channels 2 + } + ] + bindings [ + { slave 0 channel 0 } + { slave 0 channel 1 } + { slave 1 channel 0 } + { slave 1 channel 1 } + { slave 2 channel 1 } + ] +} + +<confdir:pcm/surround51.conf> +Echo_Echo3G.pcm.surround51.0 { + @args [ CARD ] + @args.CARD { + type string + } + type multi + slaves [ + { + pcm { + @func concat + strings [ + "cards.Echo_Echo3G.pcm.front.0:CARD=" $CARD + ] + } + channels 2 + } + { + pcm { + @func concat + strings [ + "cards.Echo_Echo3G.pcm.rear.0:CARD=" $CARD + ] + } + channels 2 + } + { + pcm { + @func concat + strings [ + "cards.Echo_Echo3G.pcm.center_lfe.0:CARD=" $CARD + ] + } + channels 2 + } + ] + bindings [ + { slave 0 channel 0 } + { slave 0 channel 1 } + { slave 1 channel 0 } + { slave 1 channel 1 } + { slave 2 channel 0 } + { slave 2 channel 1 } + ] +} + +<confdir:pcm/surround71.conf> +Echo_Echo3G.pcm.surround71.0 { + @args [ CARD ] + @args.CARD { + type string + } + type multi + slaves [ + { + pcm { + @func concat + strings [ + "cards.Echo_Echo3G.pcm.front.0:CARD=" $CARD + ] + } + channels 2 + } + { + pcm { + @func concat + strings [ + "cards.Echo_Echo3G.pcm.rear.0:CARD=" $CARD + ] + } + channels 2 + } + { + pcm { + @func concat + strings [ + "cards.Echo_Echo3G.pcm.center_lfe.0:CARD=" $CARD + ] + } + channels 2 + } + { + pcm { + @func concat + strings [ + "cards.Echo_Echo3G.pcm.side.0:CARD=" $CARD + ] + } + channels 2 + } + ] + bindings [ + { slave 0 channel 0 } + { slave 0 channel 1 } + { slave 1 channel 0 } + { slave 1 channel 1 } + { slave 2 channel 0 } + { slave 2 channel 1 } + { slave 3 channel 0 } + { slave 3 channel 1 } + ] +} + +<confdir:pcm/iec958.conf> +Echo_Echo3G.pcm.iec958.0 { + @args [ CARD AES0 AES1 AES2 AES3 ] + @args.CARD { + type string + } + @args.AES0 { + type integer + } + @args.AES1 { + type integer + } + @args.AES2 { + type integer + } + @args.AES3 { + type integer + } + type hooks + slave.pcm { + type hw + card $CARD + device 1 + } + + hooks.0 { + type ctl_elems + hook_args [ + { + interface PCM + name "IEC958 Playback Default" + preserve true + optional true + value [ $AES0 $AES1 $AES2 $AES3 ] + } + ] + } +} diff --git a/src/conf/cards/Makefile.am b/src/conf/cards/Makefile.am index b7190e7..ee7991b 100644 --- a/src/conf/cards/Makefile.am +++ b/src/conf/cards/Makefile.am @@ -19,6 +19,7 @@ cfg_files = aliases.conf \ CMI8738-MC8.conf \ CMI8788.conf \ CS46xx.conf \ + Echo_Echo3G.conf \ EMU10K1.conf \ EMU10K1X.conf \ ENS1370.conf \ -- 1.7.9

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[alsa-devel] Request for help CM109 on arm board
by Jonathan Andrews 07 Feb '12

07 Feb '12

Hi all, I've written an application on the PC that works with alsa, it uses 8 mono 22050Hz 16 bit audio channels via alsa. I'm trying to port the application to an Arm board with only a CM109 via USB. I have a working alsa library, I can play raw audio via the aplay utility and record via arecord ok. My problem is I don't understand the alsa configuration in enough detail to get it right. At the moment I have an alsa.conf cribbed from google, seems to work but only as a passthough. # cat /usr/share/alsa/alsa.conf_old pcm.!default { type hw card 0 } ctl.!default { type hw card 0 } ARM / # lsmod snd_usb_audio 65715 0 - Live 0xbf082000 snd_hwdep 4756 1 snd_usb_audio, Live 0xbf07b000 snd_pcm 54351 1 snd_usb_audio, Live 0xbf062000 snd_timer 14870 1 snd_pcm, Live 0xbf058000 snd_page_alloc 3336 1 snd_pcm, Live 0xbf052000 snd_usbmidi_lib 14853 1 snd_usb_audio, Live 0xbf048000 snd_rawmidi 14657 1 snd_usbmidi_lib, Live 0xbf03d000 snd_seq_device 4675 1 snd_rawmidi, Live 0xbf036000 cm109 6943 0 - Live 0xbf011000 dm9601 5583 0 - Live 0xbf00a000 usbnet 11888 1 dm9601, Live 0xbf000000 ARM / # cat /proc/asound/cards 0 [default ]: USB-Audio - Generic USB Audio Device Generic USB Audio Device at usb-ep93xx-3, full speed ARM / # cat /proc/asound/devices 0: [ 0] : control 16: [ 0- 0]: digital audio playback 24: [ 0- 0]: digital audio capture 33: : timer ARM / # I do not have enough of a machine or distribution to use the alsaconf utility. Could someone help me with an alsa.conf that will do what roughly what I am after : IE to allow an application to open N channels each 22050Hz 16 bit mono to be mixed and resampled for an output via a CM109 that only provides 44.1 or 48Khz 16 bit interleaved ... phew ... Thanks, Jon

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[alsa-devel] [PATCH 1/2] ASoC: Convert kirkwood-t5325 to use snd_soc_register_card()
by Axel Lin 07 Feb '12

07 Feb '12

Use snd_soc_register_card() instead of creating a "soc-audio" platform device. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- I don't have this hardware, I'd appreciate if someone can test this patch. Axel arch/arm/mach-kirkwood/t5325-setup.c | 6 ++++ sound/soc/kirkwood/kirkwood-t5325.c | 47 +++++++++++++++++---------------- 2 files changed, 30 insertions(+), 23 deletions(-) diff --git a/arch/arm/mach-kirkwood/t5325-setup.c b/arch/arm/mach-kirkwood/t5325-setup.c index 966b2b3..f9d2a11 100644 --- a/arch/arm/mach-kirkwood/t5325-setup.c +++ b/arch/arm/mach-kirkwood/t5325-setup.c @@ -106,6 +106,11 @@ static struct platform_device hp_t5325_button_device = { } }; +static struct platform_device hp_t5325_audio_device = { + .name = "t5325-audio", + .id = -1, +}; + static unsigned int hp_t5325_mpp_config[] __initdata = { MPP0_NF_IO2, MPP1_SPI_MOSI, @@ -179,6 +184,7 @@ static void __init hp_t5325_init(void) kirkwood_sata_init(&hp_t5325_sata_data); kirkwood_ehci_init(); platform_device_register(&hp_t5325_button_device); + platform_device_register(&hp_t5325_audio_device); i2c_register_board_info(0, i2c_board_info, ARRAY_SIZE(i2c_board_info)); kirkwood_audio_init(); diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c index 16b83e5..ce4f408 100644 --- a/sound/soc/kirkwood/kirkwood-t5325.c +++ b/sound/soc/kirkwood/kirkwood-t5325.c @@ -80,7 +80,6 @@ static struct snd_soc_dai_link t5325_dai[] = { }, }; - static struct snd_soc_card t5325 = { .name = "t5325", .owner = THIS_MODULE, @@ -93,38 +92,40 @@ static struct snd_soc_card t5325 = { .num_dapm_routes = ARRAY_SIZE(t5325_route), }; -static struct platform_device *t5325_snd_device; - -static int __init t5325_init(void) +static int __devinit t5325_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &t5325; int ret; - if (!machine_is_t5325()) - return 0; - - t5325_snd_device = platform_device_alloc("soc-audio", -1); - if (!t5325_snd_device) - return -ENOMEM; - - platform_set_drvdata(t5325_snd_device, - &t5325); - - ret = platform_device_add(t5325_snd_device); - if (ret) { - printk(KERN_ERR "%s: platform_device_add failed\n", __func__); - platform_device_put(t5325_snd_device); - } + card->dev = &pdev->dev; + ret = snd_soc_register_card(card); + if (ret) + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); return ret; } -module_init(t5325_init); -static void __exit t5325_exit(void) +static int __devexit t5325_remove(struct platform_device *pdev) { - platform_device_unregister(t5325_snd_device); + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + return 0; } -module_exit(t5325_exit); + +static struct platform_driver t5325_driver = { + .driver = { + .name = "t5325-audio", + .owner = THIS_MODULE, + }, + .probe = t5325_probe, + .remove = __devexit_p(t5325_remove), +}; + +module_platform_driver(t5325_driver); MODULE_AUTHOR("Arnaud Patard <arnaud.patard(a)rtp-net.org>"); MODULE_DESCRIPTION("ALSA SoC t5325 audio client"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:t5325-audio"); -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: wm8994: Fix typo in VMID ramp setting
by Mark Brown 07 Feb '12

07 Feb '12

The VMID ramp rate is supposed to be 3b, not 0x11. Fix that. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 81795eb..b75a652 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -785,7 +785,7 @@ static void vmid_reference(struct snd_soc_codec *codec) WM8994_VMID_RAMP_MASK, WM8994_STARTUP_BIAS_ENA | WM8994_VMID_BUF_ENA | - (0x11 << WM8994_VMID_RAMP_SHIFT)); + (0x3 << WM8994_VMID_RAMP_SHIFT)); /* Main bias enable, VMID=2x40k */ snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, -- 1.7.9.rc1

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[alsa-devel] [RESUBMIT][PATCH v2 0/4] Amstrad Delta: access MODEM_RESET GPIO pin over a regulator
by Janusz Krzysztofik 07 Feb '12

07 Feb '12

The Amstrad Delta on-board latch2 bit named MODEM_NRESET, now available as a GPIO pin AMS_DELTA_GPIO_PIN_NMODEM_RESET, is used to power up/down (bring into/out of a reset state) two distinct on-board devices simultaneously: the modem, and the voice codec. As a consequence, that bit is, or can be, manipulated concurrently by two drivers, or their platform provided hooks. Instead of updating those drivers to use the gpiolib API as a new method of controlling the MODEM_NRESET pin state, like it was done to other drivers accessing latch2 pins, and still being vulnerable to potential concurrency conflicts, or trying to solve that sharing issue with a custom piece of code, set up a fixed regulator device on top of that GPIO pin and update both drivers to manipulate that regulator, not the GPIO pin directly. I'm resubmitting v2 of this series, originally posted on 30 Dec 2011, with Mark's Ack already collected, now rebased on top of current linux-omap/omap1 tip, commit 967809bd7faf71ddc29c8081e0f21db8b201a0f4. Since patch 2/4, "ASoC: cx20442: add bias control over a platform provided regulator", has already been applied by Liam (thanks!) and is already present in mainline, I've omitted it from this submission. v2 changes against initial version: * in both the codec and the modem callbacks, don't track the regulator enable/disable state, compare new target bias level (the codec case) or power state (the modem case) with the old value instead; thanks to Mark Brown who suggested this solution, * a few other minor changes, mostly stylistic. Janusz Krzysztofik (3): ARM: OMAP1: ams-delta: set up regulator over modem reset GPIO pin ARM: OMAP1: ams-delta: update the modem to use regulator API ASoC: OMAP: ams-delta: drop .set_bias_level callback arch/arm/mach-omap1/Kconfig | 2 + arch/arm/mach-omap1/board-ams-delta.c | 94 +++++++++++++++++++-- arch/arm/plat-omap/include/plat/board-ams-delta.h | 1 - sound/soc/omap/ams-delta.c | 32 ------- 4 files changed, 87 insertions(+), 42 deletions(-) -- 1.7.3.4

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Re: [alsa-devel] linux-next: Tree for Feb 7 (sound/soc)
by Randy Dunlap 07 Feb '12

07 Feb '12

On 02/06/2012 10:09 PM, Stephen Rothwell wrote: > Hi all, > > Changes since 20120206: > > > The sound-asoc tree lost its build failure. on x86_64: ERROR: "regmap_update_bits_check" [sound/soc/snd-soc-core.ko] undefined! Full randconfig file is attached. -- ~Randy *** Remember to use Documentation/SubmitChecklist when testing your code ***

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Re: [alsa-devel] [PATCH v3] ASoC: Add support for TI LM49453 Audio codec
by Mark Brown 07 Feb '12

07 Feb '12

On Mon, Feb 06, 2012 at 06:20:26AM -0800, Reddy, MR Swami wrote: > Changes made in v3: > o Updated the lm49453_set_dai_pll() as per review comments in v2 patch. > o Removed pll disable code in _set_dai_pll(). This doesn't really seem to address the issue at all - you still have the problems with the set_pll() function not doing anything it's supposed to do with the input and output frequencies, and now there's no way to disable the PLL.

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[alsa-devel] [PATCH] ASoC: wm_hubs: Convert most output drivers to OUT_DRV widgets
by Mark Brown 07 Feb '12

07 Feb '12

No practical impact but now we have the control type we may as well use it for the slightly nicer sequencing. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_hubs.c | 24 ++++++++++++------------ 1 files changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index e469483..a53daf6 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -656,10 +656,10 @@ SND_SOC_DAPM_MIXER("SPKR Boost", SND_SOC_NOPM, 0, 0, right_speaker_boost, ARRAY_SIZE(right_speaker_boost)), SND_SOC_DAPM_SUPPLY("TSHUT", WM8993_POWER_MANAGEMENT_2, 14, 0, NULL, 0), -SND_SOC_DAPM_PGA("SPKL Driver", WM8993_POWER_MANAGEMENT_1, 12, 0, - NULL, 0), -SND_SOC_DAPM_PGA("SPKR Driver", WM8993_POWER_MANAGEMENT_1, 13, 0, - NULL, 0), +SND_SOC_DAPM_OUT_DRV("SPKL Driver", WM8993_POWER_MANAGEMENT_1, 12, 0, + NULL, 0), +SND_SOC_DAPM_OUT_DRV("SPKR Driver", WM8993_POWER_MANAGEMENT_1, 13, 0, + NULL, 0), SND_SOC_DAPM_MIXER("LINEOUT1 Mixer", SND_SOC_NOPM, 0, 0, line1_mix, ARRAY_SIZE(line1_mix)), @@ -675,14 +675,14 @@ SND_SOC_DAPM_MIXER("LINEOUT2N Mixer", SND_SOC_NOPM, 0, 0, SND_SOC_DAPM_MIXER("LINEOUT2P Mixer", SND_SOC_NOPM, 0, 0, line2p_mix, ARRAY_SIZE(line2p_mix)), -SND_SOC_DAPM_PGA("LINEOUT1N Driver", WM8993_POWER_MANAGEMENT_3, 13, 0, - NULL, 0), -SND_SOC_DAPM_PGA("LINEOUT1P Driver", WM8993_POWER_MANAGEMENT_3, 12, 0, - NULL, 0), -SND_SOC_DAPM_PGA("LINEOUT2N Driver", WM8993_POWER_MANAGEMENT_3, 11, 0, - NULL, 0), -SND_SOC_DAPM_PGA("LINEOUT2P Driver", WM8993_POWER_MANAGEMENT_3, 10, 0, - NULL, 0), +SND_SOC_DAPM_OUT_DRV("LINEOUT1N Driver", WM8993_POWER_MANAGEMENT_3, 13, 0, + NULL, 0), +SND_SOC_DAPM_OUT_DRV("LINEOUT1P Driver", WM8993_POWER_MANAGEMENT_3, 12, 0, + NULL, 0), +SND_SOC_DAPM_OUT_DRV("LINEOUT2N Driver", WM8993_POWER_MANAGEMENT_3, 11, 0, + NULL, 0), +SND_SOC_DAPM_OUT_DRV("LINEOUT2P Driver", WM8993_POWER_MANAGEMENT_3, 10, 0, + NULL, 0), SND_SOC_DAPM_OUTPUT("SPKOUTLP"), SND_SOC_DAPM_OUTPUT("SPKOUTLN"), -- 1.7.9.rc1

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[alsa-devel] [patch] ALSA: hda ca0132 - remove unused error handling code
by Dan Carpenter 07 Feb '12

07 Feb '12

We never save the return value from chipio_write() so "err" is always zero here, not less than zero. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- The other option would be to save the return value. I don't have the hardware and I worried that there was a reason we don't save it. The current code is misleading because it looks like it's checked until you notice that the "err = " bit is missing. diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 35abe3c..92f1f0c 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -733,8 +733,6 @@ static int ca0132_hp_switch_put(struct snd_kcontrol *kcontrol, /* *valp 0 is mute, 1 is unmute */ data = (data & 0x7f) | (*valp ? 0 : 0x80); chipio_write(codec, REG_CODEC_MUTE, data); - if (err < 0) - return err; spec->curr_hp_switch = *valp; @@ -775,8 +773,6 @@ static int ca0132_speaker_switch_put(struct snd_kcontrol *kcontrol, /* *valp 0 is mute, 1 is unmute */ data = (data & 0xef) | (*valp ? 0 : 0x10); chipio_write(codec, REG_CODEC_MUTE, data); - if (err < 0) - return err; spec->curr_speaker_switch = *valp; @@ -824,14 +820,10 @@ static int ca0132_hp_volume_put(struct snd_kcontrol *kcontrol, val = 31 - left_vol; data = (data & 0xe0) | val; chipio_write(codec, REG_CODEC_HP_VOL_L, data); - if (err < 0) - return err; val = 31 - right_vol; data = (data & 0xe0) | val; chipio_write(codec, REG_CODEC_HP_VOL_R, data); - if (err < 0) - return err; spec->curr_hp_volume[0] = left_vol; spec->curr_hp_volume[1] = right_vol;

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