April 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Mixer problem
by James Gardiner 05 Apr '09

05 Apr '09

I'm currently in the process of adding support for my HP dv4-1222nr laptop to the drivers. It has an IDT 92HD71B7 codec chip and no matter how hard I try, the mixer wants to default to the ATI SB one. I'm checking my /proc and dmesg and it's telling me the same thing. The weird thing is that using "model=hp-dv5" in my modprobe is fixing the 0x0d node speaker problem and my mods to get the digital microphone operative are working somewhat. However, the mixer in the sigmatel source shows that it's capable of adding dmic boost to the mixer controls. A printk I added in the source at the point where it would start processing the dmic is not even reaching the function. I tried adding my subsystem id to the PCI quirks and that didn't even help. I'm lost... Any ideas?

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[alsa-devel] 0 or EAGAIN from snd_pcm_update_avail()?
by Ludovico Cavedon 05 Apr '09

05 Apr '09

Hi all, if snd_pcm_update_avail() gets called on a PCM device in non-blocking mode when no data is availabe jet, what is the expected return value? -0 -EAGAIN -either is valid I am asking this question because pulseaudio expect to receive 0, but the the bluetooth headset pcm transfer callback returns EAGAIN, so I want to find out against whoms I should open a bug :) Thanks in advance, Ludovico

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[alsa-devel] how to use mcbsp for i2s?
by Ernesto Torres 04 Apr '09

04 Apr '09

I'm currently working in adding an external codec to the beagleboard, so far the i2c control part is functional, now i want to use mcbsp3 to provide i2s transfers Is there any way to test the mcbsp to validate the connection? Also, I've done little modifications to the omap3beagle.c to change from mcbsp2 to mcbsp3, can someone validate this is correct? In: static int __init omap3beagle_soc_init(void) I changed to: *(unsigned int *)omap3beagle_dai.cpu_dai->private_data = 2; /* McBSP3 */ It was: *(unsigned int *)omap3beagle_dai.cpu_dai->private_data = 1; /* McBSP2 */ Is this correct? Or it should be changed in other places too? Thanks

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[alsa-devel] bug report
by Jeremy 04 Apr '09

04 Apr '09

I was told to post this as a bug report I dont have mic boost on my creative blaster Audigy SE . I am in Debian Squeezey. Here is my alsa info from the script: http://www.alsa-project.org/db/?f=a6a639d7543e25379e5ed2e0e9e3fbab35d285cb there is actually another mic on my webcam that is usb, if you are wondering what that is. Thanks, Jeremy

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[alsa-devel] [PATCH] ASoC: TWL4030: Add actuall support for 96KHz playback support
by Peter Ujfalusi 03 Apr '09

03 Apr '09

Adds the needed code to be able to use 96KHz playback. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/twl4030.c | 3 +++ sound/soc/codecs/twl4030.h | 1 + 2 files changed, 4 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 4199498..bfda7a8 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1311,6 +1311,9 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, case 48000: mode |= TWL4030_APLL_RATE_48000; break; + case 96000: + mode |= TWL4030_APLL_RATE_96000; + break; default: printk(KERN_ERR "TWL4030 hw params: unknown rate %d\n", params_rate(params)); diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index 33dbb14..cb63765 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -109,6 +109,7 @@ #define TWL4030_APLL_RATE_32000 0x80 #define TWL4030_APLL_RATE_44100 0x90 #define TWL4030_APLL_RATE_48000 0xA0 +#define TWL4030_APLL_RATE_96000 0xE0 #define TWL4030_SEL_16K 0x04 #define TWL4030_CODECPDZ 0x02 #define TWL4030_OPT_MODE 0x01 -- 1.6.2

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[alsa-devel] [PATCH][TRIVIAL] remove extra space
by Viral Mehta 03 Apr '09

03 Apr '09

Just for the sake of readability, removing extra space Signed-off-by: Viral Mehta <viral.mehta(a)einfochips.com> --- --- a/sound/usb/usbmixer.c 2009-03-24 04:42:14.000000000 +0530 +++ b/sound/usb/usbmixer.c 2009-04-03 12:49:49.000000000 +0530 @@ -394,7 +394,7 @@ static int set_ctl_value(struct usb_mixe value_set = convert_bytes_value(cval, value_set); buf[0] = value_set & 0xff; buf[1] = (value_set >> 8) & 0xff; - while (timeout -- > 0) + while (timeout-- > 0) if (snd_usb_ctl_msg(cval->mixer->chip->dev, usb_sndctrlpipe(cval->mixer->chip->dev, 0), request,

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[alsa-devel] Missing pcmC0D0p on Intel HDA with Alsa 20090327 snapshot
by Frederik Himpe 03 Apr '09

03 Apr '09

Hi, I'm using Mandriva kernel 2.6.29-tmb-desktop-2mdv, which according to the changelog contains Alsa 1.0.19+ snapshot 20090327 and Alsa 1.0.19+ unstable addons snapshot 20090327. With this kernel, I have no sound working on my Intel HDA device on a Dell Latitude E6400 system, while sound is working fine with standard 2.6.29 kernel. aplay gives this output: ALSA lib pcm_dmix.c:1008:(snd_pcm_dmix_open) unable to open slave aplay: main:590: audio open error: No such file or directory According to strace, this is because of a missing /dev/snd/pcmC0D0p. # ls /dev/snd controlC0 hwC0D0 hwC0D2 seq timer Some dmesg related output: HDA Intel 0000:00:1b.0: PCI INT A -> GSI 21 (level, low) -> IRQ 21 HDA Intel 0000:00:1b.0: setting latency timer to 64 input: HDA Digital PCBeep as /class/input/input9 hda_codec: formats == 0 (nid=0x10, val=0xe07e0, ovrd=0, streams=0xe07e0) hda_codec: cannot attach PCM stream 0 for codec #0 hda_codec: formats == 0 (nid=0x2, val=0x1e07f0, ovrd=1, streams=0x1e07f0) hda_codec: cannot attach PCM stream 3 for codec #2 input: HDA Intel Mic at Sep Left Jack as /class/input/input10 input: HDA Intel Mic at Ext Right Jack as /class/input/input11 input: HDA Intel Line Out at Sep Left Jack as /class/input/input12 input: HDA Intel HP Out at Ext Right Jack as /class/input/input13 lspci: 00:1b.0 Audio device [0403]: Intel Corporation 82801I (ICH9 Family) HD Audio Controller [8086:293e] (rev 03) # lsmod | grep snd snd_hda_codec_intelhdmi 13856 1 snd_hda_codec_idt 69004 1 snd_hda_intel 29128 0 snd_hda_codec 68864 3 snd_hda_codec_intelhdmi,snd_hda_codec_idt,snd_hda_intel snd_hwdep 8488 1 snd_hda_codec snd_seq_dummy 3140 0 snd_seq_oss 31456 0 snd_seq_midi_event 7296 1 snd_seq_oss snd_seq 56704 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi_event snd_seq_device 7156 3 snd_seq_dummy,snd_seq_oss,snd_seq snd_pcm_oss 43392 0 snd_pcm 80520 3 snd_hda_intel,snd_hda_codec,snd_pcm_oss snd_timer 22832 2 snd_seq,snd_pcm snd_mixer_oss 16512 1 snd_pcm_oss snd 65192 12 snd_hda_codec_intelhdmi,snd_hda_codec_idt,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_seq_oss,snd_seq,snd_seq_device,snd_pcm_oss,snd_pcm,snd_timer,snd_mixer_oss soundcore 7648 1 snd snd_page_alloc 9680 2 snd_hda_intel,snd_pcm -- Frederik Himpe

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[alsa-devel] [PATCH 3/3] ARM: DaVinci: ASoC: Add machine driver support for DM646x
by Naresh Medisetty 03 Apr '09

03 Apr '09

This patch does the following: (1) Add support for the DM646x machine (2)Introduce the platform driver model to get platform data for all the machines including dm355 and dm644x. Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git sound/soc/davinci/davinci-evm.c | 129 ++++++++++++++++++++++++++++++++------- 1 files changed, 107 insertions(+), 22 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 9b90b34..b89a322 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -20,11 +20,19 @@ #include <sound/soc-dapm.h> #include <asm/dma.h> -#include <mach/hardware.h> +#include <asm/mach-types.h> + +#include <mach/asp.h> +#include <mach/edma.h> +#include <mach/mux.h> +#include <mach/cpu.h> + #include "../codecs/tlv320aic3x.h" +#include "../codecs/davinci_codec_stubs.h" #include "davinci-pcm.h" #include "davinci-i2s.h" +#include "davinci-i2s-mcasp.h" #define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \ @@ -39,7 +47,7 @@ static int evm_hw_params(struct snd_pcm_substream *substream, unsigned sysclk; /* ASP1 on DM355 EVM is clocked by an external oscillator */ - if (machine_is_davinci_dm355_evm()) + if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm()) sysclk = 27000000; /* ASP0 in DM6446 EVM is clocked by U55, as configured by @@ -140,6 +148,25 @@ static struct snd_soc_dai_link evm_dai = { .ops = &evm_ops, }; +static struct snd_soc_dai_link dm6467_evm_dai[] = { + { + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai = davinci_iis_mcasp_dai, + .codec_dai = &aic3x_dai, + .init = evm_aic3x_init, + .ops = &evm_ops, + }, + { + .name = "MCASP SPDIF", + .stream_name = "spdif", + .cpu_dai = &davinci_iis_mcasp_dai[1], + .codec_dai = dit_stub_dai, + .ops = &evm_ops, + }, +}; + + /* davinci-evm audio machine driver */ static struct snd_soc_card snd_soc_card_evm = { .name = "DaVinci EVM", @@ -148,12 +175,27 @@ static struct snd_soc_card snd_soc_card_evm = { .num_links = 1, }; +/* davinci dm6467 evm audio machine driver */ +static struct snd_soc_card dm6467_snd_soc_card_evm = { + .name = "DaVinci DM6467 EVM", + .platform = &davinci_soc_platform, + .dai_link = dm6467_evm_dai, + .num_links = 2, +}; + /* evm audio private data */ static struct aic3x_setup_data evm_aic3x_setup = { .i2c_bus = 0, .i2c_address = 0x1b, }; +/* dm6467 evm audio private data */ +static struct aic3x_setup_data dm6467_evm_aic3x_setup = { + .i2c_bus = 1, + .i2c_address = 0x18, +}; + + /* evm audio subsystem */ static struct snd_soc_device evm_snd_devdata = { .card = &snd_soc_card_evm, @@ -161,36 +203,58 @@ static struct snd_soc_device evm_snd_devdata = { .codec_data = &evm_aic3x_setup, }; -static struct resource evm_snd_resources[] = { - { - .start = DAVINCI_MCBSP_BASE, - .end = DAVINCI_MCBSP_BASE + SZ_8K - 1, - .flags = IORESOURCE_MEM, - }, -}; - -static struct evm_snd_platform_data evm_snd_data = { - .tx_dma_ch = DM644X_DMACH_MCBSP_TX, - .rx_dma_ch = DM644X_DMACH_MCBSP_RX, +/* evm audio subsystem */ +static struct snd_soc_device dm6467_evm_snd_devdata = { + .card = &dm6467_snd_soc_card_evm, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &dm6467_evm_aic3x_setup, }; static struct platform_device *evm_snd_device; -static int __init evm_init(void) +static int davinci_evm_probe(struct platform_device *pdev) { + struct resource *resources; + struct evm_snd_platform_data *data = pdev->dev.platform_data; + struct snd_soc_device *evm_snd_dev_data; + int index; int ret; + int res_size = pdev->num_resources; + + if (!data) + return -ENODEV; + + resources = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!resources) + return -ENODEV; + + if (machine_is_davinci_evm()) { + davinci_cfg_reg(DM644X_MCBSP); + evm_snd_dev_data = &evm_snd_devdata; + index = 0; + } else if (machine_is_davinci_dm355_evm()) { + /* we don't use ASP1 IRQs, or we'd need to mux them ... */ + davinci_cfg_reg(DM355_EVT8_ASP1_TX); + davinci_cfg_reg(DM355_EVT9_ASP1_RX); + evm_snd_dev_data = &evm_snd_devdata; + index = 1; + } else if (machine_is_davinci_dm6467_evm()) { + evm_snd_dev_data = &dm6467_evm_snd_devdata; + index = 0; + } else + return -EINVAL; - evm_snd_device = platform_device_alloc("soc-audio", 0); + evm_snd_device = platform_device_alloc("soc-audio", index); if (!evm_snd_device) return -ENOMEM; - platform_set_drvdata(evm_snd_device, &evm_snd_devdata); - evm_snd_devdata.dev = &evm_snd_device->dev; - platform_device_add_data(evm_snd_device, &evm_snd_data, - sizeof(evm_snd_data)); + platform_set_drvdata(evm_snd_device, evm_snd_dev_data); + evm_snd_dev_data->dev = &evm_snd_device->dev; + evm_snd_device->dev.platform_data = data; + + ret = platform_device_add_resources(evm_snd_device, + resources, res_size); - ret = platform_device_add_resources(evm_snd_device, evm_snd_resources, - ARRAY_SIZE(evm_snd_resources)); if (ret) { platform_device_put(evm_snd_device); return ret; @@ -203,11 +267,32 @@ static int __init evm_init(void) return ret; } -static void __exit evm_exit(void) +static int davici_evm_remove(struct platform_device *pdev) { platform_device_unregister(evm_snd_device); + return 0; +} +static struct platform_driver davinci_asoc_driver = { + .probe = davinci_evm_probe, + .remove = davici_evm_remove, + .driver = { + .name = "davinci-asoc", + .owner = THIS_MODULE, + }, +}; + +static int __init evm_init(void) +{ + return platform_driver_register(&davinci_asoc_driver); } +static void __exit evm_exit(void) +{ + platform_driver_unregister(&davinci_asoc_driver); +} + + + module_init(evm_init); module_exit(evm_exit); -- 1.5.6

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[alsa-devel] [PATCH 2/5] ARM: DaVinci: ASoC: Adds McASP clock support for TI DM646X processor
by naresh＠ti.com 03 Apr '09

03 Apr '09

From: Naresh Medisetty <naresh(a)ti.com> Adds McASP clock support for the two instances of mcasp (mcasp0,mcasp1). This patch is part of the audio support for dm646x series. Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies to the DaVinci git tree available at http://git.kernel.org/?p=linux/kernel/git/khilman/linux-davinci.git arch/arm/mach-davinci/dm646x.c | 14 ++++++++++++++ 1 files changed, 14 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-davinci/dm646x.c b/arch/arm/mach-davinci/dm646x.c index fbc9daf..6276b31 100644 --- a/arch/arm/mach-davinci/dm646x.c +++ b/arch/arm/mach-davinci/dm646x.c @@ -220,6 +220,18 @@ static struct clk gpio_clk = { .lpsc = DM646X_LPSC_GPIO, }; +static struct clk mcasp0_clk = { + .name = "mcasp0", + .parent = &pll1_sysclk3, + .lpsc = DM646X_LPSC_McASP0, +}; + +static struct clk mcasp1_clk = { + .name = "mcasp1", + .parent = &pll1_sysclk3, + .lpsc = DM646X_LPSC_McASP1, +}; + static struct clk aemif_clk = { .name = "aemif", .parent = &pll1_sysclk3, @@ -307,6 +319,8 @@ struct davinci_clk dm646x_clks[] = { CLK(NULL, "uart2", &uart2_clk), CLK("i2c_davinci.1", NULL, &i2c_clk), CLK(NULL, "gpio", &gpio_clk), + CLK(NULL, "mcasp0", &mcasp0_clk), + CLK(NULL, "mcasp1", &mcasp1_clk), CLK(NULL, "aemif", &aemif_clk), CLK("davinci_emac.1", NULL, &emac_clk), CLK(NULL, "pwm0", &pwm0_clk), -- 1.5.6

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[alsa-devel] iMac 9,1 sound patch
by Micah Bucy 02 Apr '09

02 Apr '09

I tried a git snapshot, but it wasn't working at all, so this patch was developed against the 1.0.19 stable release. This is my first time delving into alsa, so I'm not sure if everything is how it should be, including if the patch was correctly created. I can tell you this. I was able to get jack detection working correctly as long as the 6 channel option is set. I had tried modifying it to only allow a 6ch option as 2ch is for line-out (headphones) only, but the internal speakers wouldn't work, so I left it alone. My guess is, this same setup could be used by the macbook pro 5,1 I'll give it a try over the weekend. I have not tested the built-in mic or the line-in. -Micah

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