May 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [RFC PATCH 0/5] OMAP/ASoC: McBSP: FIFO handling related fixes
by Peter Ujfalusi 01 Jun '10

01 Jun '10

Hello, This series aims to correct how the McBSP FIFO is viewed, and handled. Introduction of the problem: OMAP McBSP FIFO is word structured: McBSP2 has 1024 + 256 = 1280 word long buffer, McBSP1,3,4,5 has 128 word long buffer This means, that the size of the FIFO depends on the McBSP word size configuration. For example on McBSP3: 16bit samples: size is 128 * 2 = 256 bytes 32bit samples: size is 128 * 4 = 512 bytes It is simpler to place constraint for buffer and period based on channels. McBSP3 as example again (16 or 32 bit samples): 1 channel (mono): size is 128 frames (128 words) 2 channels (stereo): size is 128 / 2 = 64 frames (2 * 64 words) 4 channels: size is 128 / 4 = 32 frames (4 * 32 words) Since now the McBSP codec supports not only 16bit samples (32biut has been recently added), the FIFO size handling is no longer correct, since it has been hard wired for 16bit word length. The series changes how the users of McBSP are configuring the FIFO: It used to be 0 based (0 meant 1 word threshold). After this series users can configure the threshold in 1 base mode (1 means 1 word threshold). The platform code now provides the _full_ size of the FIFO in words, instead of the already limited value used in the past. In ASoC omap-mcbsp code hw_rule based constraint refinement is going to be used instead of the hardwired static constraint, which was correct only in case of 16bit word length. The hw_rule is refining the minimum buffer size based on the channel number going to be used by the coming stream. In case of threshold mode additional hw_rule refines the maximum allowed period size. The series are generated agains Takashi's sound-2.6: topic/asoc branch. CCing also Eduardo, and Eero since they have worked on the original FIFO/threshold implementation. All commetns and testers are welcome! Peter --- Peter Ujfalusi (5): OMAP: McBSP: Function to query the FIFO size OMAP3: McBSP: Change the way how the FIFO is handled OMAP3: McBSP: Use the port's buffer_size when calculating tx delay ASoC: omap-mcbsp: Save, and use wlen for threshold configuration ASoC: omap-mcbsp: Place correct constraints for streams arch/arm/mach-omap2/mcbsp.c | 10 ++-- arch/arm/plat-omap/include/plat/mcbsp.h | 2 + arch/arm/plat-omap/mcbsp.c | 31 ++++++--- sound/soc/omap/omap-mcbsp.c | 112 +++++++++++++++++++++++------- 4 files changed, 114 insertions(+), 41 deletions(-)

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Re: [alsa-devel] Problem with snd_hda_intel from 05-27-2010 snapshot
by Jaroslav Kysela 31 May '10

31 May '10

On Sun, 30 May 2010, VDR User wrote: > On Sun, May 30, 2010 at 8:37 AM, Jaroslav Kysela <perex(a)perex.cz> wrote: >> Thanks. The wallclock register seems working OK. Please, use value 29 for >> the xrun_debug file to do more accurate inspection what's going wrong (as I >> suggested in my first reply): > > Attached is my full syslog with xrun_debug 29. Thanks. Could you try this patch? diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 170610e..ac3dbab 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1910,7 +1910,7 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev) if (WARN_ONCE(!azx_dev->period_bytes, "hda-intel: zero azx_dev->period_bytes")) return -1; /* this shouldn't happen! */ - if (wallclk <= azx_dev->period_wallclk && + if (wallclk < (azx_dev->period_wallclk * 3) / 2 && pos % azx_dev->period_bytes > azx_dev->period_bytes / 2) /* NG - it's below the first next period boundary */ return bdl_pos_adj[chip->dev_index] ? 0 : -1; ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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[alsa-devel] [PATCH 1/9] ALSA: usb-audio: UAC2: clean up parsing of bmaControls
by Daniel Mack 31 May '10

31 May '10

Introduce two new static inline functions for a more readable parsing of UAC2 bmaControls. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- include/linux/usb/audio-v2.h | 10 ++++++++++ sound/usb/mixer.c | 8 ++++---- 2 files changed, 14 insertions(+), 4 deletions(-) diff --git a/include/linux/usb/audio-v2.h b/include/linux/usb/audio-v2.h index 92f1d99..8f91be3 100644 --- a/include/linux/usb/audio-v2.h +++ b/include/linux/usb/audio-v2.h @@ -18,6 +18,16 @@ /* v1.0 and v2.0 of this standard have many things in common. For the rest * of the definitions, please refer to audio.h */ +static inline bool uac2_control_is_readable(u32 bmControls, u8 control) +{ + return (bmControls >> (control * 2)) & 0x1; +} + +static inline bool uac2_control_is_writeable(u32 bmControls, u8 control) +{ + return (bmControls >> (control * 2)) & 0x2; +} + /* 4.7.2.1 Clock Source Descriptor */ struct uac_clock_source_descriptor { diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 03ce971..43d6417 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1188,9 +1188,9 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void for (j = 0; j < channels; j++) { unsigned int mask = snd_usb_combine_bytes(bmaControls + csize * (j+1), csize); - if (mask & (1 << (i * 2))) { + if (uac2_control_is_readable(mask, i)) { ch_bits |= (1 << j); - if (~mask & (1 << ((i * 2) + 1))) + if (!uac2_control_is_writeable(mask, i)) ch_read_only |= (1 << j); } } @@ -1198,9 +1198,9 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void /* FIXME: the whole unit is read-only if any of the channels is marked read-only */ if (ch_bits & 1) /* the first channel must be set (for ease of programming) */ build_feature_ctl(state, _ftr, ch_bits, i, &iterm, unitid, !!ch_read_only); - if (master_bits & (1 << i * 2)) + if (uac2_control_is_readable(master_bits, i)) build_feature_ctl(state, _ftr, 0, i, &iterm, unitid, - ~master_bits & (1 << ((i * 2) + 1))); + !uac2_control_is_writeable(master_bits, i)); } } -- 1.7.1

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[alsa-devel] [PATCH] ASoC: spdif: Add codec driver to use spdif stand-alone
by Seungwhan Youn 31 May '10

31 May '10

This patch adds spdif dummy codec driver for using spdif-dit as a stand-alone. Until this, spdif-dit can be used only with other codecs like tlv320aci3x in davinci platform. Signed-off-by: Seungwhan Youn <sw.youn(a)samsung.com> --- sound/soc/codecs/spdif_transciever.c | 93 +++++++++++++++++++++++++++++++++- 1 files changed, 92 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/spdif_transciever.c b/sound/soc/codecs/spdif_transciever.c index a631911..f0945ab 100644 --- a/sound/soc/codecs/spdif_transciever.c +++ b/sound/soc/codecs/spdif_transciever.c @@ -18,6 +18,7 @@ #include <linux/moduleparam.h> #include <sound/soc.h> #include <sound/pcm.h> +#include <sound/initval.h> #include "spdif_transciever.h" @@ -26,6 +27,48 @@ MODULE_LICENSE("GPL"); #define STUB_RATES SNDRV_PCM_RATE_8000_96000 #define STUB_FORMATS SNDRV_PCM_FMTBIT_S16_LE +static struct snd_soc_codec *spdif_dit_codec; + +static int spdif_dit_codec_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret; + + if (spdif_dit_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = spdif_dit_codec; + codec = spdif_dit_codec; + + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto err_create_pcms; + } + + return 0; + +err_create_pcms: + return ret; +} + +static int spdif_dit_codec_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_spdif_dit = { + .probe = spdif_dit_codec_probe, + .remove = spdif_dit_codec_remove, +}; EXPORT_SYMBOL_GPL(soc_codec_dev_spdif_dit); + struct snd_soc_dai dit_stub_dai = { .name = "DIT", .playback = { @@ -40,13 +83,61 @@ EXPORT_SYMBOL_GPL(dit_stub_dai); static int spdif_dit_probe(struct platform_device *pdev) { + struct snd_soc_codec *codec; + int ret; + + if (spdif_dit_codec) { + dev_err(&pdev->dev, "Another Codec is registered\n"); + ret = -EINVAL; + goto err_reg_codec; + } + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + codec->dev = &pdev->dev; + + mutex_init(&codec->mutex); + + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->name = "spdif-dit"; + codec->owner = THIS_MODULE; + codec->dai = &dit_stub_dai; + codec->num_dai = 1; + + spdif_dit_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto err_reg_codec; + } + dit_stub_dai.dev = &pdev->dev; - return snd_soc_register_dai(&dit_stub_dai); + ret = snd_soc_register_dai(&dit_stub_dai); + if (ret < 0) { + dev_err(codec->dev, "Failed to register dai: %d\n", ret); + goto err_reg_dai; + } + + return 0; + +err_reg_dai: + snd_soc_unregister_codec(codec); +err_reg_codec: + kfree(spdif_dit_codec); + return ret; } static int spdif_dit_remove(struct platform_device *pdev) { snd_soc_unregister_dai(&dit_stub_dai); + snd_soc_unregister_codec(spdif_dit_codec); + kfree(spdif_dit_codec); + spdif_dit_codec = NULL; return 0; } -- 1.6.2.5

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[alsa-devel] i.MX SSI Kconfig fixes
by Sascha Hauer 31 May '10

31 May '10

Hi, Two small fixes for i.MX SSI. Sascha The following changes since commit 67a3e12b05e055c0415c556a315a3d3eb637e29e: Linux 2.6.35-rc1 (2010-05-30 13:21:02 -0700) are available in the git repository at: git://git.pengutronix.de/git/imx/linux-2.6.git imx-ssi-fixes Sascha Hauer (2): i.MX ssi: MX31ads sound support should depend on MACH_MX31ADS i.MX ssi: Add missing Kconfig entry for Phytec boards sound/soc/imx/Kconfig | 11 ++++++++++- 1 files changed, 10 insertions(+), 1 deletions(-)

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[alsa-devel] ASoC updates for 2.6.36
by Mark Brown 31 May '10

31 May '10

The following changes since commit 67a3e12b05e055c0415c556a315a3d3eb637e29e: Linux 2.6.35-rc1 (2010-05-30 13:21:02 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.36 This includes a merge up of 2.6.35-rc1, plus the commits below. Barry Song (1): ASoC: ad193x: add set_sysclk entry to support different clock input Jassi Brar (2): ASOC: S5PC100: Enable AC97 support ASOC: S5PV210: Enable AC97 support Mark Brown (7): Merge branch 'topic/asoc' of git://git.kernel.org/.../tiwai/sound-2.6 into for-2.6.36 ASoC: Remove version display from WM8990 ASoC: Fix dB scales for WM835x ASoC: Fix dB scales for WM8400 ASoC: Fix dB scales for WM8990 Merge commit 'v2.6.35-rc1' into for-2.6.36 ASoC: Fix S/PDIF build Seungwhan Youn (1): ASoC: spdif: Add codec driver to use spdif stand-alone Wan ZongShun (1): ASoC: NUC900/audio: add nuc900 audio driver support apatard(a)mandriva.com (7): ASoC: Add SOC_DOUBLE_R_SX_TLV control ASoC: Add SND_SOC_DAPM_PRE_POST_PMD event ASoC: Add driver for cs42l51 orion/kirkwood: add audio functions. openrd-client: initialise audio ASoC: kirkwood: Add i2s support ASoC: kirkwood: Add audio support to openrd client platforms arch/arm/mach-kirkwood/common.c | 38 ++ arch/arm/mach-kirkwood/common.h | 2 + arch/arm/mach-kirkwood/include/mach/kirkwood.h | 3 + arch/arm/mach-kirkwood/openrd-setup.c | 13 + arch/arm/plat-orion/include/plat/audio.h | 11 + include/sound/soc-dapm.h | 2 + include/sound/soc.h | 21 + sound/soc/Kconfig | 2 + sound/soc/Makefile | 2 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ad193x.c | 41 ++- sound/soc/codecs/ad193x.h | 5 + sound/soc/codecs/cs42l51.c | 763 ++++++++++++++++++++++++ sound/soc/codecs/cs42l51.h | 163 +++++ sound/soc/codecs/spdif_transciever.c | 94 +++- sound/soc/codecs/spdif_transciever.h | 1 + sound/soc/codecs/wm8990.c | 4 - sound/soc/kirkwood/Kconfig | 20 + sound/soc/kirkwood/Makefile | 9 + sound/soc/kirkwood/kirkwood-dma.c | 383 ++++++++++++ sound/soc/kirkwood/kirkwood-dma.h | 17 + sound/soc/kirkwood/kirkwood-i2s.c | 485 +++++++++++++++ sound/soc/kirkwood/kirkwood-i2s.h | 17 + sound/soc/kirkwood/kirkwood-openrd.c | 126 ++++ sound/soc/kirkwood/kirkwood.h | 126 ++++ sound/soc/nuc900/Kconfig | 27 + sound/soc/nuc900/Makefile | 11 + sound/soc/nuc900/nuc900-ac97.c | 442 ++++++++++++++ sound/soc/nuc900/nuc900-audio.c | 81 +++ sound/soc/nuc900/nuc900-auido.h | 121 ++++ sound/soc/nuc900/nuc900-pcm.c | 352 +++++++++++ sound/soc/s3c24xx/Kconfig | 4 +- sound/soc/s3c24xx/smdk_wm9713.c | 3 + sound/soc/soc-core.c | 95 +++ 35 files changed, 3482 insertions(+), 8 deletions(-) create mode 100644 arch/arm/plat-orion/include/plat/audio.h create mode 100644 sound/soc/codecs/cs42l51.c create mode 100644 sound/soc/codecs/cs42l51.h create mode 100644 sound/soc/kirkwood/Kconfig create mode 100644 sound/soc/kirkwood/Makefile create mode 100644 sound/soc/kirkwood/kirkwood-dma.c create mode 100644 sound/soc/kirkwood/kirkwood-dma.h create mode 100644 sound/soc/kirkwood/kirkwood-i2s.c create mode 100644 sound/soc/kirkwood/kirkwood-i2s.h create mode 100644 sound/soc/kirkwood/kirkwood-openrd.c create mode 100644 sound/soc/kirkwood/kirkwood.h create mode 100644 sound/soc/nuc900/Kconfig create mode 100644 sound/soc/nuc900/Makefile create mode 100644 sound/soc/nuc900/nuc900-ac97.c create mode 100644 sound/soc/nuc900/nuc900-audio.c create mode 100644 sound/soc/nuc900/nuc900-auido.h create mode 100644 sound/soc/nuc900/nuc900-pcm.c

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[alsa-devel] [GIT PULL] ASoC updates for 2.6.36
by Liam Girdwood 31 May '10

31 May '10

Hi Takashi, Please pull the following ASoC updates for 2.6.36. Thanks Liam --- The following changes since commit ad8332c1302bcb4f80d593fd3eb477be9d7f5604: Jarkko Nikula (1): ASoC: tpa6130a2: Remove CPVSS and HPVdd supplies are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git for-2.6.36 Jorge Eduardo Candelaria (1): ASoC: TWL6040: Fix playback with 19.2 Mhz MCLK Peter Ujfalusi (8): ASoC: TWL4030: Revisit codec defaults ASoC: TWL4030: Remove wrapper for power down ASoC: TWL4030: Make offset cancellation path configurable ASoC: TWL4030: Optimize the power up sequence ASoC: TWL4030: Helper to check chip default registers ASoC: TWL4030: Correct the ARXR2_APGA_CTL chip default ASoC: TWL4030: Use BIAS_OFF instead of BIAS_STANDBY, when not in use ASoC: TWL4030: Add functionalty to reset the registers sound/soc/codecs/twl4030.c | 291 +++++++++++++++++++++++++------------------- sound/soc/codecs/twl4030.h | 3 + sound/soc/codecs/twl6040.c | 2 +- 3 files changed, 170 insertions(+), 126 deletions(-)

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[alsa-devel] alsa-driver-1.0.23 fails to build on ppc
by Elimar Riesebieter 31 May '10

31 May '10

Hi, preparing alsa-driver-1.0.23 for Debian upload I found while building modules all: LD [M] /usr/src/modass/usr_src/modules/alsa-driver/aoa/core/snd-aoa.o CC [M] /usr/src/modass/usr_src/modules/alsa-driver/aoa/fabrics/layout.o In file included from /usr/src/modass/usr_src/modules/alsa-driver/aoa/fabrics/layout.c:2: /usr/src/modass/usr_src/modules/alsa-driver/aoa/fabrics/../../alsa-kernel/aoa/fabrics/layout.c: In function 'check_codec': /usr/src/modass/usr_src/modules/alsa-driver/aoa/fabrics/../../alsa-kernel/aoa/fabrics/layout.c:772: error: 'struct device_node' has no member named 'phandle' make[8]: *** [/usr/src/modass/usr_src/modules/alsa-driver/aoa/fabrics/layout.o] Error 1 make[7]: *** [/usr/src/modass/usr_src/modules/alsa-driver/aoa/fabrics] Error 2 make[6]: *** [/usr/src/modass/usr_src/modules/alsa-driver/aoa] Error 2 make[5]: *** [_module_/usr/src/modass/usr_src/modules/alsa-driver] Error 2 make[4]: *** [sub-make] Error 2 make[3]: *** [all] Error 2 I found no fix in git. I would be glad if someone could give me a hint. Thanks Elimar -- Planung: Ersatz des Zufalls durch den Irrtum. -unknown-

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Re: [alsa-devel] [PATCH] alsa/Au1xxx-PSC: use resource_size
by Manuel Lauss 31 May '10

31 May '10

2010/5/28 Wan ZongShun <mcuos.com(a)gmail.com>: > Dear Manuel & Mark, > > I make a little modification to use the 'resource_size'. It's okay with me. You could also modifiy psc-ac97.c in the same way. Manuel Lauss

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[alsa-devel] [patch 0/6] kirkwood openrd client audio support - v4
by apatard＠mandriva.com 31 May '10

31 May '10

Hi, This patchset is an other try to get merged openrd audio support mainline. The main change is going back to using kirkwood instead of orion, as the only orion having i2s is not merged mainline (if I got it right). I've also fixed the lastests comments on the cs42l51 codec patch. Arnaud Changes : - changed back from orion to kirkwood - removed cs42l51 adc input register initialisation - moved the DAPM event macro to soc-dapm - fixed cs42l51_set_dai_fmt() master/slave check - removed cs42l51 startup message - use snd_soc_update_bits() in cs42l51_pdn_event()

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[alsa-devel] arecord and DMA misaligned error with device
by Ruchi Sirauthiya 31 May '10

31 May '10

Hi all, I am working on an OMAP board with Linux kernel 2.6.22.18-omap3 and ALSA. I got the following stack trace when running 'arecord' continuously for a long time (~30 minutes). What could be the reason for "page allocation failure" and "DMA misaligned error"? How can I solve this issue? Thanks and Regards, Ruchi Sirauthiya ---- LOG START ---- page allocation failure. order:5, mode:0xd1 [ 1249.046875] [<c0050380>] (dump_stack+0x0/0x14) from [<c00ae260>] (__alloc_pages+0x288/0x29c) [ 1249.054687] [<c00adfd8>] (__alloc_pages+0x0/0x29c) from [<c0051cac>] (__dma_alloc+0x17c/0x3d4) [ 1249.062500] [<c0051b30>] (__dma_alloc+0x0/0x3d4) from [<c0051f58>] (dma_alloc_coherent+0x24/0x30) [ 1249.070312] [<c0051f34>] (dma_alloc_coherent+0x0/0x30) from [<c02717b4>] (snd_omap_alsa_hw_params+0x54/0x7c) [ 1249.085937] [<c0271760>] (snd_omap_alsa_hw_params+0x0/0x7c) from [<c02683dc>] (snd_pcm_hw_params+0x90/0x2e0) [ 1249.093750] r5:c19d6000 r4:c15da000 [ 1249.093750] [<c026834c>] (snd_pcm_hw_params+0x0/0x2e0) from [<c02686b0>] (snd_pcm_hw_params_user+0x84/0xe4) [ 1249.109375] r9:cb8f4000 r8:c004c028 r7:c077e360 r6:c19d6000 r5:c077e360 [ 1249.109375] r4:001bdb48 [ 1249.117187] [<c026862c>] (snd_pcm_hw_params_user+0x0/0xe4) from [<c02689c0>] (snd_pcm_common_ioctl1+0x1cc/0xf04) [ 1249.125000] r7:c077e360 r6:001bdb48 r5:001bdb48 r4:c25c4111 [ 1249.132812] [<c02687f4>] (snd_pcm_common_ioctl1+0x0/0xf04) from [<c0269b44>] (snd_pcm_capture_ioctl1+0x44c/0x470) [ 1249.140625] [<c02696f8>] (snd_pcm_capture_ioctl1+0x0/0x470) from [<c0269b9c>] (snd_pcm_capture_ioctl+0x34/0x38) [ 1249.148437] [<c0269b68>] (snd_pcm_capture_ioctl+0x0/0x38) from [<c00d0adc>] (do_ioctl+0x34/0x78) [ 1249.164062] [<c00d0aa8>] (do_ioctl+0x0/0x78) from [<c00d0d80>] (vfs_ioctl+0x260/0x278) [ 1249.171875] r5:001bdb48 r4:ccb9a4e0 [ 1249.171875] [<c00d0b20>] (vfs_ioctl+0x0/0x278) from [<c00d0dd8>] (sys_ioctl+0x40/0x64) [ 1249.179687] r7:ccb9a4e0 r6:c25c4111 r5:001bdb48 r4:00000039 [ 1249.187500] [<c00d0d98>] (sys_ioctl+0x0/0x64) from [<c004be80>] (ret_fast_syscall+0x0/0x2c) [ 1249.195312] r7:00000036 r6:001ec5c8 r5:413b3bd4 r4:001bdb48 [ 1249.203125] Mem-info: [ 1249.203125] DMA per-cpu: [ 1249.203125] CPU 0: Hot: hi: 90, btch: 15 usd: 66 Cold: hi: 30, btch: 7 usd: 24 [ 1249.210937] Active:10183 inactive:35469 dirty:4 writeback:0 unstable:0 [ 1249.210937] free:739 slab:5167 mapped:2540 pagetables:125 bounce:0 [ 1249.226562] DMA free:2956kB min:1888kB low:2360kB high:2832kB active:40732kB inactive:141876kB present:223520kB pages_scanned:0 all_unreclaimable? no [ 1249.242187] lowmem_reserve[]: 0 0 [ 1249.242187] DMA: 13*4kB 33*8kB 27*16kB 3*32kB 31*64kB 1*128kB 0*256kB 0*512kB 0*1024kB 0*2048kB 0*4096kB 0*8192kB = 2956kB [ 1249.257812] Free swap: 0kB [ 1249.265625] 56320 pages of RAM [ 1249.273437] 1016 free pages [ 1249.273437] 3616 reserved pages [ 1249.273437] 5167 slab pages [ 1249.281250] 41756 pages shared [ 1249.281250] 0 pages swap cached [ 1249.343750] DMA misaligned error with device 1 [ 1249.343750] DMA misaligned error with device 1 [ 1249.351562] DMA misaligned error with device 1 [ 1249.359375] DMA misaligned error with device 1 [ 1249.359375] DMA misaligned error with device 1 [ 1249.367187] DMA misaligned error with device 1 [ 1249.367187] DMA misaligned error with device 1 [ 1249.375000] DMA misaligned error with device 1 [ 1249.375000] DMA misaligned error with device 1 [ 1249.382812] DMA misaligned error with device 1 [ 1249.390625] DMA misaligned error with device 1 [ 1249.390625] DMA misaligned error with device 1 [ 1249.398437] DMA misaligned error with device 1 [ 1249.398437] DMA misaligned error with device 1 [ 1249.406250] DMA misaligned error with device 1 [ 1249.406250] DMA misaligned error with device 1 ... ... log truncated ... ... ---- LOG END ----

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[alsa-devel] [PATCH v2 0/8] ASoC: TWL4030: put the codec to off when not active
by Peter Ujfalusi 31 May '10

31 May '10

Hello, Changes since v1: - Patch 3: removed the shifting of offset cancellation value - Added register content reset functionality as suggested by Liam and Mark I have added Mark's Acked-by to patch 1-7, patch 8 is a new one, which shall address the comments regarding to the register content reset. To-do after this series: Revisit the register reset functionality to use burst write. Intro mail from the original series: The following series at the end will let the twl4030 codec to use BIAS_OFF instead of BIAS_STANDBY. The difference in power consumption is about 0.5mA. To achieve this: - The regcache has been reseted to codec default - The codec initialization has been optimized, it is no longer writes all 73 registers at startup, but only modifies few selected one. - The power related code has been cleaned up, and optimized - Support added for machine drivers to select the offset cancellation path - debug support for checking the codec default registers (machine drivers can ask for checking, but shall be disabled in production). - Added codec register reset functionality I guess that's it. The driver has been tested on a custom board with twl5031. It passed all of our internal test cases covering much of the codec features. --- Peter Ujfalusi (8): ASoC: TWL4030: Revisit codec defaults ASoC: TWL4030: Remove wrapper for power down ASoC: TWL4030: Make offset cancellation path configurable ASoC: TWL4030: Optimize the power up sequence ASoC: TWL4030: Helper to check chip default registers ASoC: TWL4030: Correct the ARXR2_APGA_CTL chip default ASoC: TWL4030: Use BIAS_OFF instead of BIAS_STANDBY, when not in use ASoC: TWL4030: Add functionalty to reset the registers sound/soc/codecs/twl4030.c | 291 +++++++++++++++++++++++++------------------- sound/soc/codecs/twl4030.h | 3 + 2 files changed, 169 insertions(+), 125 deletions(-)

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[alsa-devel] How to support software mixing on QUALCOMM 72XX without mmap?
by Berg 31 May '10

31 May '10

Hello,guys. I'm porting ASLA1.0.23 to qualcomm 72xx base on ARM11. The first instance,"aplay test.wav", works now, but the second instance should fail with "device busy" error message. I have tried plug dmix and dshare "aply -D plug:mycard test.wav", but met a fail again for the soc chip doesn't support mmap. Of course,pcm_mycard is defined in /etc/asound.conf. In my pc ,both above cmds work well . Though,the document shows that: NOTE: For ALSA 1.0.9rc2 and higher you don't need to setup dmix for analogue output. Dmix is enabled by default for soundcards which don't support hardware mixing. You still need to set it up for digital outputs. My questions are: 1)Is software mixing implemented in ALSA Driver,or in ALSA LIB? 2)What could I do to make the software mixing work? Is there any special configure which need to adjust? And when testing on pc whth HD audio, I noticed that A)when aplay instances runs WITHOUT dmix, the result of"cat /etc/one_of_my_aplay_pid/maps" should like: ... b331d000-b731e000 rw-s 00000000 00:15 862346 /dev/shm/pulse-shm-2882420837 ... B)when aplay instances runs WITH dmix,the result is: ... b7511000-b7519000 rw-s 00000000 00:0f 4560 /dev/snd/pcmC0D0p b7519000-b751a000 rw-s 81000000 00:0f 4560 /dev/snd/pcmC0D0p b751a000-b751b000 r--s 80000000 00:0f 4560 /dev/snd/pcmC0D0p ... So,the third question: 3) From A) and B),can I make a result that the mmap feacture is not necessary for the buildin software mixing ? Expect your help,please berg

3 2

[alsa-devel] [PATCH] ALSA: hda: Use LPIB for another mainboard
by Daniel Chen 31 May '10

31 May '10

Hi, Attached is a patch against my latest LPIB patches that makes the original reporter's playback and capture audible by default. Please apply. This patch is applicable to all 2.6.3x stable series. Best, -Dan

2 1

[alsa-devel] [PATCH] ALSA: hda: Use mbp31 quirk for an iMac model
by Daniel Chen 31 May '10

31 May '10

Hi, Attached is a patch against sound-2.6 master enabling audible playback and capture for the original reporter's iMac. Please apply. This patch is applicable to all 2.6.3x stable series. Best, -Dan

2 2

[alsa-devel] [PATCH] ALSA: hda: Use LPIB for an ASUS device
by Daniel Chen 31 May '10

31 May '10

Hi, Attached is a patch against sound-2.6 master that enables the original reporter to adjust capture volume through PulseAudio without the daemon crashing. This patch is applicable to all 2.6.3x stable series. Best, -Dan

2 1

[alsa-devel] [PATCH] pcm: Define G723 3-bit and 5-bit formats
by Ben Collins 31 May '10

31 May '10

pcm: Define G723 3-bit and 5-bit formats This defines the 24bps and 40bps (8khz sample rate) G.723 codec formats. They are going to be used once I submit the driver for an mpeg4/g723 compression card. I've updated the signed value to -1 as per Takashi's comments since these are non-linear formats. Signed-off-by: Ben Collins <bcollins(a)bluecherry.net> Acked-by: Takashi Iwai <tiwai(a)suse.de> diff --git a/include/sound/asound.h b/include/sound/asound.h index 0985955..62e0857 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -212,7 +212,11 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_S18_3BE ((__force snd_pcm_format_t) 41) /* in three bytes */ #define SNDRV_PCM_FORMAT_U18_3LE ((__force snd_pcm_format_t) 42) /* in three bytes */ #define SNDRV_PCM_FORMAT_U18_3BE ((__force snd_pcm_format_t) 43) /* in three bytes */ -#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_U18_3BE +#define SNDRV_PCM_FORMAT_G723_24 ((__force snd_pcm_format_t) 44) /* 8 samples in 3 bytes */ +#define SNDRV_PCM_FORMAT_G723_24_1B ((__force snd_pcm_format_t) 45) /* 1 sample in 1 byte */ +#define SNDRV_PCM_FORMAT_G723_40 ((__force snd_pcm_format_t) 46) /* 8 Samples in 5 bytes */ +#define SNDRV_PCM_FORMAT_G723_40_1B ((__force snd_pcm_format_t) 47) /* 1 sample in 1 byte */ +#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_G723_40_1B #ifdef SNDRV_LITTLE_ENDIAN #define SNDRV_PCM_FORMAT_S16 SNDRV_PCM_FORMAT_S16_LE diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 8b611a5..37b5c3c 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -173,6 +173,10 @@ struct snd_pcm_ops { #define SNDRV_PCM_FMTBIT_U18_3LE (1ULL << SNDRV_PCM_FORMAT_U18_3LE) #define SNDRV_PCM_FMTBIT_S18_3BE (1ULL << SNDRV_PCM_FORMAT_S18_3BE) #define SNDRV_PCM_FMTBIT_U18_3BE (1ULL << SNDRV_PCM_FORMAT_U18_3BE) +#define SNDRV_PCM_FMTBIT_G723_24 (1ULL << SNDRV_PCM_FORMAT_G723_24) +#define SNDRV_PCM_FMTBIT_G723_24_1B (1ULL << SNDRV_PCM_FORMAT_G723_24_1B) +#define SNDRV_PCM_FMTBIT_G723_40 (1ULL << SNDRV_PCM_FORMAT_G723_40) +#define SNDRV_PCM_FMTBIT_G723_40_1B (1ULL << SNDRV_PCM_FORMAT_G723_40_1B) #ifdef SNDRV_LITTLE_ENDIAN #define SNDRV_PCM_FMTBIT_S16 SNDRV_PCM_FMTBIT_S16_LE diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index ea2bf82..434af3c 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -128,6 +128,14 @@ static struct pcm_format_data pcm_formats[SNDRV_PCM_FORMAT_LAST+1] = { .width = 4, .phys = 4, .le = -1, .signd = -1, .silence = {}, }, + [SNDRV_PCM_FORMAT_G723_24] = { + .width = 3, .phys = 3, .le = -1, .signd = -1, + .silence = {}, + }, + [SNDRV_PCM_FORMAT_G723_40] = { + .width = 5, .phys = 5, .le = -1, .signd = -1, + .silence = {}, + }, /* FIXME: the following three formats are not defined properly yet */ [SNDRV_PCM_FORMAT_MPEG] = { .le = -1, .signd = -1, @@ -186,6 +194,14 @@ static struct pcm_format_data pcm_formats[SNDRV_PCM_FORMAT_LAST+1] = { .width = 18, .phys = 24, .le = 0, .signd = 0, .silence = { 0x02, 0x00, 0x00 }, }, + [SNDRV_PCM_FORMAT_G723_24_1B] = { + .width = 3, .phys = 8, .le = -1, .signd = -1, + .silence = {}, + }, + [SNDRV_PCM_FORMAT_G723_40_1B] = { + .width = 5, .phys = 8, .le = -1, .signd = -1, + .silence = {}, + }, }; -- Bluecherry: http://www.bluecherrydvr.com/ SwissDisk : http://www.swissdisk.com/ Ubuntu : http://www.ubuntu.com/ My Blog : http://ben-collins.blogspot.com/

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[alsa-devel] [PATCH 1/4] snd-usb-caiaq: Restore 'Control vinyl' input mode on A4DJ
by Mark Hills 31 May '10

31 May '10

This feature was undocumented on early A4DJ units. It is indicated by lighting both the 'line' and 'phono' lamps at the same time. Newer units document this and the newer Windows drivers enable this for all units, so restore the functionality. This patch simplifies the code and changes the mode mapping to match the A8DJ, favouring simpler code and consistency over keeping the existing mapping. Both 'Control vinyl' and 'Phono' input modes enable the hardware preamp. The difference is the input impedance. This reverts commit 9a9527e. Acked-by: Daniel Mack <daniel(a)caiaq.de> Signed-off-by: Mark Hills <mark(a)pogo.org.uk> --- sound/usb/caiaq/control.c | 30 ++---------------------------- 1 files changed, 2 insertions(+), 28 deletions(-) diff --git a/sound/usb/caiaq/control.c b/sound/usb/caiaq/control.c index 36ed703..70c3866 100644 --- a/sound/usb/caiaq/control.c +++ b/sound/usb/caiaq/control.c @@ -42,21 +42,12 @@ static int control_info(struct snd_kcontrol *kcontrol, switch (dev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ): - if (pos == 0) { - /* current input mode of A8DJ */ - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 2; - return 0; - } - break; - case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ): if (pos == 0) { - /* current input mode of A4DJ */ + /* current input mode of A8DJ and A4DJ */ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; + uinfo->value.integer.max = 2; return 0; } break; @@ -86,14 +77,6 @@ static int control_get(struct snd_kcontrol *kcontrol, struct snd_usb_caiaqdev *dev = caiaqdev(chip->card); int pos = kcontrol->private_value; - if (dev->chip.usb_id == - USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ)) { - /* A4DJ has only one control */ - /* do not expose hardware input mode 0 */ - ucontrol->value.integer.value[0] = dev->control_state[0] - 1; - return 0; - } - if (pos & CNT_INTVAL) ucontrol->value.integer.value[0] = dev->control_state[pos & ~CNT_INTVAL]; @@ -113,15 +96,6 @@ static int control_put(struct snd_kcontrol *kcontrol, unsigned char cmd = EP1_CMD_WRITE_IO; switch (dev->chip.usb_id) { - case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ): { - /* A4DJ has only one control */ - /* do not expose hardware input mode 0 */ - dev->control_state[0] = ucontrol->value.integer.value[0] + 1; - snd_usb_caiaq_send_command(dev, EP1_CMD_WRITE_IO, - dev->control_state, sizeof(dev->control_state)); - return 1; - } - case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): cmd = EP1_CMD_DIMM_LEDS; break; -- 1.7.1

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[alsa-devel] [PATCH] ALSA: hda: Use LPIB for a Shuttle device
by Daniel Chen 31 May '10

31 May '10

Hi, Attached please find a patch against current sound-2.6 master that fixes audio anomalies and PulseAudio crashing by specifying LPIB method explicitly. Please apply. This patch is applicable to all 2.6.3x stable series. Best, -Dan

2 1

[alsa-devel] [PATCH] ALSA: usb/mixer.c: remove unused variable
by Daniel Mack 31 May '10

31 May '10

This was a development left-over. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Clemens Ladisch <clemens(a)ladisch.de> --- sound/usb/mixer.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index fd3e08d..a35dd74 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1970,8 +1970,6 @@ static void snd_usb_mixer_proc_read(struct snd_info_entry *entry, static void snd_usb_mixer_interrupt_v2(struct usb_mixer_interface *mixer, int attribute, int value, int index) { - struct usb_mixer_elem_info *info; - switch (attribute) { case UAC2_CS_CUR: snd_usb_mixer_notify_id(mixer, index); -- 1.7.1

3 3

Re: [alsa-devel] Problem with snd_hda_intel from 05-27-2010 snapshot
by Jaroslav Kysela 30 May '10

30 May '10

On Thu, 27 May 2010, VDR User wrote: > On Thu, May 27, 2010 at 12:37 AM, Jaroslav Kysela <perex(a)perex.cz> wrote: >> Check any suspicious lines in /var/log/messages and follow proc xrun_debug >> settings noted in bug above. I need at least 20 first lines after the >> playback is started. > > By this do you mean compile the drivers with pcm-xrun-debug > --with-debug=verbose --enable-verbose-procfs options? Also, is there > a way to disable all the debug logging in maybe > /etc/modprobe.d/alsa-base.conf? My system is installed on a CF and I > can't have alsa hammering it with writes is they're not needed but the > only way I know to turn the debug logging off is to not compile it in > the first place. I asked for one run, so you can recompile ALSA driver without debugging support after test. I think that the problem might be that the WALLCLK register gives wrong values for your hardware (but the Intel specification says that implementation of this register is MANDATORY). Here is a patch which might show the bug: diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 170610e..a1e7c97 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1891,6 +1891,7 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev) int stream; wallclk = azx_readl(chip, WALLCLK) - azx_dev->start_wallclk; + printk("wallclk: %u, %u\n", wallclk, azx_readl(chip, WALLCLK)); if (wallclk < (azx_dev->period_wallclk * 2) / 3) return -1; /* bogus (too early) interrupt */ >> Also, use 'aplay -Dplughw:0' from the alsa-utils package to trigger the >> playback. > > The hdmi device I'm using is 1,3: > > test:~$ aplay -l > **** List of PLAYBACK Hardware Devices **** > card 0: NVidia [HDA NVidia], device 0: ALC883 Analog [ALC883 Analog] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: NVidia [HDA NVidia], device 1: ALC883 Digital [ALC883 Digital] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 1: NVidia_1 [HDA NVidia], device 3: NVIDIA HDMI [NVIDIA HDMI] > Subdevices: 0/1 > Subdevice #0: subdevice #0 > > So what exactly do I need to do then with aplay? Do I need some file > to play? Sorry, I've never used Linux as a desktop so I'm only > familiar with running VDR from command line. If you use HDMI device, use 'aplay plughw:1,0 <some_wav_file>'. Some_wav_file means a .wav type audio file. Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

2 2

[alsa-devel] How to support software mixing on QUALCOMM 72XX without mmap?
by Berg.Tian 29 May '10

29 May '10

Hello,guys. I'm porting ASLA1.0.23 to qualcomm 72xx base on ARM11. The first instance,"aplay test.wav", works now, but the second instance should fail with "device busy" error message. I have tried plug dmix and dshare "aply -D plug:mycard test.wav", but met a fail again for the soc chip doesn't support mmap. Of course,pcm_mycard is defined in /etc/asound.conf. In my pc ,both above cmds work well . Though,the document shows that: NOTE: For ALSA 1.0.9rc2 and higher you don't need to setup dmix for analogue output. Dmix is enabled by default for soundcards which don't support hardware mixing. You still need to set it up for digital outputs. My questions are: 1)Is software mixing implemented in ALSA Driver,or in ALSA LIB? 2)What could I do to make the software mixing work? Is there any special configure which need to adjust? And when testing on pc whth HD audio, I noticed that A)when aplay instances runs WITHOUT dmix, the result of"cat /etc/one_of_my_aplay_pid/maps" should like: ... b331d000-b731e000 rw-s 00000000 00:15 862346 /dev/shm/pulse-shm-2882420837 ... B)when aplay instances runs WITH dmix,the result is: ... b7511000-b7519000 rw-s 00000000 00:0f 4560 /dev/snd/pcmC0D0p b7519000-b751a000 rw-s 81000000 00:0f 4560 /dev/snd/pcmC0D0p b751a000-b751b000 r--s 80000000 00:0f 4560 /dev/snd/pcmC0D0p ... So,the third question: 3) From A) and B),can I make a result that the mmap feacture is not necessary for the buildin software mixing ? Expect your help,please berg

1 0

Re: [alsa-devel] Mixer volume/decibel mapping
by Clemens Ladisch 29 May '10

29 May '10

Raymond Yau wrote: > 2010/5/21 Clemens Ladisch <clemens(a)ladisch.de> > > DB_LINEAR means that the mixer control value is a multiplier for the > > PCM sample value, as James wrote. A DB_LINEAR mapping has > > 0 == -infinity dB, full scale == 0 dB, and the mapping between mixer > > control values and corresponding dB values it _not_ linear. > > pci/oxygen/oxygen.c:static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0); > pci/oxygen/hifier.c:static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0); > > Refer to AK4396 datasheet > > Output Volume > > The AK4396 includes channel independent digital output volumes (ATT) with > 256 levels at linear step including MUTE. > These volumes are in front of the DAC and can attenuate the input data from > 0dB to –48dB and mute. > > dB range is -48dB to 0dB only > > ATT7-0: Attenuation Level > ATT = 20 log10 (ATT_DATA / 255) [dB] > FFH : 0dB (Default) > 00H : Mute This is the section of the datasheet on which the above declarations are based. Now if you have some remark or question, you should remember to include it in your mail. Regards, Clemens

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Re: [alsa-devel] [PATCH] ALSA/davinci: Use the resource_size , __devinit and __devexit.
by Sergei Shtylyov 28 May '10

28 May '10

Hello. Wan ZongShun wrote: > Dear all, > Use the resource_size function instead of manually calculating the > resource size. This reduces the chance of introducing off-by-one > errors. > In addition, to use __devinit and __devexit is good too. "Good" isn't a good justification I think. Also, this just asks to be put into two patches instead of one as the patch does two completely different things. > Signed-off-by:Wan ZongShun<mcuos.com(a)gmail.com> WBR, Sergei

1 0

[alsa-devel] [PATCH] ALSA, hda: Add support for another Lenovo ThinkPad Edge in conexant codec
by Andreas Herrmann 28 May '10

28 May '10

From: Andreas Herrmann <andreas.herrmann3(a)amd.com> On a Thinkpad Edge 13 "01972NG" I had the problem that speakers played sound although headphones were plugged in. Using model=ideapad with latest alsa-git kernel fixed this. So adding this quirk to use ideapad for another Thinkpad Edge variant seems sensible. Cc: Jerone Young <jerone.young(a)canonical.com> Signed-off-by: Andreas Herrmann <andreas.herrmann3(a)amd.com> --- sound/pci/hda/patch_conexant.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) Patch is against today's sound-2.6 git tree. Please apply. Thanks, Andreas diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index f4a2bd6..2bf2cb5 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -2975,6 +2975,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x17aa, 0x21b2, "Thinkpad X100e", CXT5066_IDEAPAD), + SND_PCI_QUIRK(0x17aa, 0x21b3, "Thinkpad Edge 13 (197)", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x21b4, "Thinkpad Edge", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), -- 1.7.1

2 1