November 2008 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] azx_probe() cleanup
by Wu Fengguang 20 Nov '08

20 Nov '08

Replace 5 free-and-return-err blocks with goto-out-free ones. This makes the main logic more outstanding. Signed-off-by: Wu Fengguang <wfg(a)linux.intel.com> --- sound/pci/hda/hda_intel.c | 33 +++++++++++++-------------------- 1 file changed, 13 insertions(+), 20 deletions(-) --- sound-2.6.orig/sound/pci/hda/hda_intel.c +++ sound-2.6/sound/pci/hda/hda_intel.c @@ -2317,40 +2317,30 @@ static int __devinit azx_probe(struct pc } err = azx_create(card, pci, dev, pci_id->driver_data, &chip); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err < 0) + goto out_free; card->private_data = chip; /* create codec instances */ err = azx_codec_create(chip, model[dev], probe_mask[dev]); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err < 0) + goto out_free; /* create PCM streams */ err = snd_hda_build_pcms(chip->bus); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err < 0) + goto out_free; /* create mixer controls */ err = azx_mixer_create(chip); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err < 0) + goto out_free; snd_card_set_dev(card, &pci->dev); err = snd_card_register(card); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err < 0) + goto out_free; pci_set_drvdata(pci, card); chip->running = 1; @@ -2359,6 +2349,9 @@ static int __devinit azx_probe(struct pc dev++; return err; +out_free: + snd_card_free(card); + return err; } static void __devexit azx_remove(struct pci_dev *pci)

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[alsa-devel] [PATCH] ASoC: DaVinci: Fix audio stall when doing full duplex
by naresh＠ti.com 20 Nov '08

20 Nov '08

From: Naresh Medisetty <naresh(a)ti.com> Fix concurrent capture/playback issue. The issue is caused by re-initialization of control registers used specifically for capture or playback in both capture and playback operations. Signed-off-by: Steve Chen <schen(a)mvista.com> Naresh Medisetty <naresh(a)ti.com> --- sound/soc/davinci/davinci-i2s.c | 31 +++++++++++++++++++------------ 1 files changed, 19 insertions(+), 12 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 8c1bf87..11c20d0 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -295,10 +295,14 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, u32 w; /* general line settings */ - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, - DAVINCI_MCBSP_SPCR_RINTM(3) | - DAVINCI_MCBSP_SPCR_XINTM(3) | - DAVINCI_MCBSP_SPCR_FREE); + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + w |= DAVINCI_MCBSP_SPCR_RINTM(3) | DAVINCI_MCBSP_SPCR_FREE; + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + } else { + w |= DAVINCI_MCBSP_SPCR_XINTM(3) | DAVINCI_MCBSP_SPCR_FREE; + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + } i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SRGR_REG); @@ -329,16 +333,19 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_RCR_REG); - MOD_REG_BIT(w, DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) | - DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length), 1); - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, w); + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_RCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) | + DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length), 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, w); - w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_XCR_REG); - MOD_REG_BIT(w, DAVINCI_MCBSP_XCR_XWDLEN1(mcbsp_word_length) | - DAVINCI_MCBSP_XCR_XWDLEN2(mcbsp_word_length), 1); - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, w); + } else { + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_XCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_XCR_XWDLEN1(mcbsp_word_length) | + DAVINCI_MCBSP_XCR_XWDLEN2(mcbsp_word_length), 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, w); + } return 0; } -- 1.5.4.1

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[alsa-devel] PCM3008 ALSA Soc driver and Lyrtech SFFSDR board audio support (v2)
by Hugo Villeneuve 19 Nov '08

19 Nov '08

Reworked and splitted patch based on Mark Brown comments. Mark, thank-you for your comments. I implemented almost all of your suggestions, except for the de-emphasis control because I don't really know how I can export (and test) that interface. Like you said, maybe we or someone else can do it later. Hugo Villeneuve

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[alsa-devel] [PATCH 0/6] ASoC updates
by Mark Brown 19 Nov '08

19 Nov '08

The following changes since commit 72f2b894455775b980a5ac7ae70ab560b3d3d247: Mark Brown (1): ASoC: Move uda134x_codec.h to uda134x.h are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai Arun KS (1): ASoC: Add support for omap2evm board Hugo Villeneuve (2): ASoC: Add PCM3008 ALSA SoC driver ASoC: Add driver for the Lyrtech SFFSDR board Mark Brown (3): ASoC: Remove unused snd_soc_machine_config declaration ASoC: s3c24xx_uda134x DAI accessor functions and static cleanup ASoC: Convert blackfin machines to use DAI accessor functions include/sound/soc.h | 1 - sound/soc/blackfin/bf5xx-ad73311.c | 2 +- sound/soc/blackfin/bf5xx-ssm2602.c | 6 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/pcm3008.c | 201 +++++++++++++++++++++++++++++++++++ sound/soc/codecs/pcm3008.h | 25 +++++ sound/soc/davinci/Kconfig | 10 ++ sound/soc/davinci/Makefile | 2 + sound/soc/davinci/davinci-sffsdr.c | 156 +++++++++++++++++++++++++++ sound/soc/omap/Kconfig | 9 ++ sound/soc/omap/Makefile | 2 + sound/soc/omap/omap2evm.c | 150 ++++++++++++++++++++++++++ sound/soc/s3c24xx/s3c24xx_uda134x.c | 27 +++--- 14 files changed, 578 insertions(+), 19 deletions(-) create mode 100644 sound/soc/codecs/pcm3008.c create mode 100644 sound/soc/codecs/pcm3008.h create mode 100644 sound/soc/davinci/davinci-sffsdr.c create mode 100644 sound/soc/omap/omap2evm.c

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[alsa-devel] omap3evm asoc
by naveen krishna ch 19 Nov '08

19 Nov '08

Hello Alsa devels, Is there any previous work on audio ASOC for OMAP3EVM. I have started working on OMAP3EVM + TWL4030 combination and audio is working. -- Thanks, (: Naveen Krishna Ch :)

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[alsa-devel] [PATCH 1/1] ASoC: Add support for omap2evm board
by Arun KS 19 Nov '08

19 Nov '08

This patch adds twl4030 audio support on omap2evm Signed-off-by: Arun KS <arunks(a)mistralsolutions.com> --- sound/soc/omap/Kconfig | 9 +++ sound/soc/omap/Makefile | 2 + sound/soc/omap/omap2evm.c | 150 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 161 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/omap2evm.c diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index cf40e42..6c56277 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -30,4 +30,13 @@ config SND_OMAP_SOC_OVERO help Say Y if you want to add support for SoC audio on the Gumstix Overo. +config SND_OMAP_SOC_OMAP2EVM + tristate "SoC Audio support for OMAP2EVM board" + depends on SND_OMAP_SOC && MACH_OMAP2EVM + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on the omap2evm board. + + diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index fefc9be..f5da3cc 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -9,8 +9,10 @@ obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o snd-soc-n810-objs := n810.o snd-soc-osk5912-objs := osk5912.o snd-soc-overo-objs := overo.o +snd-soc-omap2evm-objs := omap2evm.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o +obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o diff --git a/sound/soc/omap/omap2evm.c b/sound/soc/omap/omap2evm.c new file mode 100644 index 0000000..c376213 --- /dev/null +++ b/sound/soc/omap/omap2evm.c @@ -0,0 +1,150 @@ +/* + * omap2evm.c -- SoC audio machine driver for omap2evm board + * + * Author: Arun KS <arunks(a)mistralsolutions.com> + * + * Based on sound/soc/omap/overo.c by Steve Sakoman + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <mach/gpio.h> +#include <mach/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/twl4030.h" + +static int omap2evm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops omap2evm_ops = { + .hw_params = omap2evm_hw_params, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link omap2evm_dai = { + .name = "TWL4030", + .stream_name = "TWL4030", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai, + .ops = &omap2evm_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_machine snd_soc_machine_omap2evm = { + .name = "omap2evm", + .dai_link = &omap2evm_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device omap2evm_snd_devdata = { + .machine = &snd_soc_machine_omap2evm, + .platform = &omap_soc_platform, + .codec_dev = &soc_codec_dev_twl4030, +}; + +static struct platform_device *omap2evm_snd_device; + +static int __init omap2evm_soc_init(void) +{ + int ret; + + if (!machine_is_omap2evm()) { + pr_debug("Not omap2evm!\n"); + return -ENODEV; + } + printk(KERN_INFO "omap2evm SoC init\n"); + + omap2evm_snd_device = platform_device_alloc("soc-audio", -1); + if (!omap2evm_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(omap2evm_snd_device, &omap2evm_snd_devdata); + omap2evm_snd_devdata.dev = &omap2evm_snd_device->dev; + *(unsigned int *)omap2evm_dai.cpu_dai->private_data = 1; /* McBSP2 */ + + ret = platform_device_add(omap2evm_snd_device); + if (ret) + goto err1; + + return 0; + +err1: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(omap2evm_snd_device); + + return ret; +} +module_init(omap2evm_soc_init); + +static void __exit omap2evm_soc_exit(void) +{ + platform_device_unregister(omap2evm_snd_device); +} +module_exit(omap2evm_soc_exit); + +MODULE_AUTHOR("Arun KS <arunks(a)mistralsolutions.com>"); +MODULE_DESCRIPTION("ALSA SoC omap2evm"); +MODULE_LICENSE("GPL"); -- 1.5.6.3

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[alsa-devel] Old OMAP mailing list
by Mark Brown 19 Nov '08

19 Nov '08

Please don't CC any ASoC patches to the linux-omap-open-source list - my understanding is that it should no longer be used, having been replaced by linux-omap(a)vger.kernel.org. Any posts CCed to it result in backscatter like this: On Tue, Nov 18, 2008 at 07:57:55AM -0600, linux-omap-open-source-owner(a)linux.omap.com wrote: > You are not allowed to post to this mailing list, and your message has > been automatically rejected. If you think that your messages are > being rejected in error, contact the mailing list owner at > linux-omap-open-source-owner(a)linux.omap.com. which seems a bit hostile. It'd be nice if this bounce mentioned the new list, FWIW.

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Re: [alsa-devel] [RFC][PATCH] ELD routines and proc interface
by Wu Fengguang 19 Nov '08

19 Nov '08

On Tue, Nov 18, 2008 at 11:55:45PM -0800, Shane W wrote: > On Wed, Nov 19, 2008 at 03:17:57PM +0800, Wu Fengguang wrote: > > On Tue, Nov 18, 2008 at 11:11:35PM -0800, Shane W wrote: > > > On Thu, Nov 13, 2008 at 10:21:53AM +0800, Wu Fengguang wrote: > > > > Create hda_eld.c for ELD routines and proc interface. > > > > > Just testing this and your multichannel HDMI code over here > > > but it doesn't seem to get ELD data for my setup. Intel > > > g45-id board with an Yamaha RX-v1800 receiver which is HDMI > > > 1.3 capable. > > > > Do you run the latest xorg intel driver? Also dmesg should tell us > > something if anything goes wrong with ELD retrieving/parsing. > > Yes though Xorg wasn't running when I was doing the audio > tests. Pure console at that point but I fired up Xorg > which is latest git as of today but still no ELD Yeah sorry I missed one note. The xorg git tree only contains the audio output enabling patches, but not yet ELD transferring patches. (the ELD data flow is monitor => video driver => audio driver) I called for the Xorg ELD enabling patches from Ma Ling just after replying to your email. He should send the patches soon. > population. When Xorg starts, I get this in dmesg: > [drm] Initialized i915 1.6.0 20080730 on minor 0 > [drm:i915_gem_object_bind_to_gtt] *ERROR* GTT full, but LRU > list empty > [drm:i915_gem_object_pin] *ERROR* Failure to bind: > -12<6>HDMI intrinsic event: PD=1 ELDV=1 This last dmesg line is funny. The ELDV(i.e. ELD Valid) is true, but the video driver obviously has no ELD data prepared for us. Thanks, Fengguang

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[alsa-devel] [PATCH 0/5] assorted HDMI cleanups
by Wu Fengguang 19 Nov '08

19 Nov '08

Hi Takashi, Here are 5 HDMI cleanup patches, some renames the code and others renames the user visible output messages. The new output goes like: dmesg [ 5742.169578] detected monitor <9d><80>t connection type Display Port [ 5742.169580] available speakers: FL/FR LFE FLC/FRC [ 5742.169582] supports coding type MLP (Dolby TrueHD): channels = 7, rates = 48000 176400 [ 5742.169585] supports coding type AC-3: channels = 1, rates = 48000 88200 192000, max bitrate = 1208000 [ 5742.169588] supports coding type AC-3: channels = 2, rates = 48000 384000, max bitrate = 960000 [ 5742.169591] supports coding type ATRAC: channels = 5, rates = 48000 88200 384000, max bitrate = 224000 /proc/asound/card0/eld\#3 monitor name �� connection_type Display Port eld_version [0xb] reserved edid_version [0x3] CEA-861-B, C or D manufacture_id 0xee5a product_id 0x923c port_id 0x18b263a03e17d1b5 support_hdcp 1 support_ai 1 audio_sync_delay 216 speakers [0x23] FL/FR LFE FLC/FRC sad_count 4 sad0_coding_type [0xc] MLP (Dolby TrueHD) sad0_channels 7 sad0_rates [0x240] 48000 176400 sad1_coding_type [0x2] AC-3 sad1_channels 1 sad1_rates [0x4c0] 48000 88200 192000 sad1_max_bitrate 1208000 sad2_coding_type [0x2] AC-3 sad2_channels 2 sad2_rates [0x1840] 48000 384000 sad2_max_bitrate 960000 sad3_coding_type [0x8] ATRAC sad3_channels 5 sad3_rates [0x8c0] 48000 88200 384000 sad3_max_bitrate 224000 Thanks, Fengguang --

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[alsa-devel] Listing up to 8 PCM devices
by Gregoire ETIENNE 19 Nov '08

19 Nov '08

Hi, I'm currently writting a driver with up to 8 pcm devices, so I turned on "Dynamic device file minor numbers" option. But when I try to list devices through aplay or a record (which are calling ioctl with cmd SNDRV_CTL_IOCTL_PCM_NEXT_DEVICE) I can only list devices from 0 to 7. This is due to this code : linux/sound/core/pcm.c l.64 ------------------------------ case SNDRV_CTL_IOCTL_PCM_NEXT_DEVICE: { int device; if (get_user(device, (int __user *)arg)) return -EFAULT; mutex_lock(&register_mutex); device = device < 0 ? 0 : device + 1; while (device < SNDRV_PCM_DEVICES) { if (snd_pcm_search(card, device)) break; device++; } if (device == SNDRV_PCM_DEVICES) device = -1; mutex_unlock(&register_mutex); if (put_user(device, (int __user *)arg)) return -EFAULT; return 0; } ------------------------------ because SNDRV_PCM_DEVICES is defined to 8 and do not change even if I selected "Dynamic device file minor numbers" option. Is there anything I missed ?

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