April 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Per-host .asoundrc on shared $HOME
by Mark Hills 23 Apr '13

23 Apr '13

There is a very old thread [1] on alsa-user on this topic which seems to go through some yucky solutions. I have the same use case, and post some findings here. Since pre-2003 alsa-lib calls wordexp(3) on the filenames. I am using the patch below in /usr/share/alsa/alsa.conf It follows a precedent set by other files such as .Xdefaults-hostname. I'd submit a patch, except there seems to be no guarantee that HOSTNAME is in the environment (on Slackware it is explicitly set in /etc/profile.) Without it I can only see a bash local variable. >From the old thread I imagine this function (or at least this mail) would be useful to others. Would it be better to patch the C code to provide a token (eg. '\h') that is substituted by the hostname? Or is it enough to assume that HOSTNAME would be set? I didn't consider short vs. long hostnames, I suppose $HOSTNAME gives the user/admin some control over this. This: ~/.asoundrc-$(hostname -s) may work, but I don't see that forking a process is an option. [1] http://thread.gmane.org/gmane.linux.alsa.user/27532/focus=27542 -- Mark --- alsa.conf.orig 2013-04-23 21:22:45.105454493 +0100 +++ alsa.conf 2013-04-23 21:16:41.707053308 +0100 @@ -17,6 +17,7 @@ } "/etc/asound.conf" "~/.asoundrc" + "~/.asoundrc-$HOSTNAME" ] errors false }

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[alsa-devel] [PATCH 00/21] ARM build fixes for 3.10
by Arnd Bergmann 23 Apr '13

23 Apr '13

Hi everyone, This is a set of bug fixes for build errors and warnings I am seeing in the various defconfigs and/or allmodconfig builds on linux-next. The majority of these showed up because we now include a lot more platforms in allmodconfig than we used to. Please apply to appropriate maintainer trees or provide feedback, I can take care of the rest sending those through arm-soc that I get no feedback on. Arnd Arnd Bergmann (21): Turn off -Wmaybe-uninitialized when building with -Os ARM: compressed/head.S: work around new binutils warning ARM: use optimized do_div only for EABI ARM: kvm: define KVM_ARM_MAX_VCPUS unconditionally ARM: OMAP2+: add dependencies on ARCH_MULTI_V6/V7 ARM: tegra: call cpu_do_idle from C code ARM: tegra: unify tegra_idle_device definitions ASoC: ux500: forward declare msp_i2s_platform_data spi/sirf: fix MODULE_DEVICE_TABLE ssb: fix alignment of struct bcma_device_id drm/tilcdc: use only a single module device table mmc: sdhci-tegra: fix MODULE_DEVICE_TABLE misc: mark spear13xx-pcie-gadget as broken staging: nvec: allow modular build staging: comedi/dawboard2000: use mdelay for large delays staging/zsmalloc: don't use pgtable-mapping from modules tty/serial/sirf: fix MODULE_DEVICE_TABLE usb: phy: tegra: don't call into tegra-ehci directly usb: phy: don't mark usb_bind_phy as __init MFD: ab8500: export ab8500_gpadc_sw_hw_convert properly mtd: omap2: allow bulding as a module Makefile | 2 +- arch/arm/boot/compressed/Makefile | 2 +- arch/arm/boot/compressed/head.S | 12 ++++++++++++ arch/arm/include/asm/div64.h | 2 +- arch/arm/kvm/Kconfig | 6 +++--- arch/arm/mach-omap2/Kconfig | 5 +++++ arch/arm/mach-tegra/cpuidle-tegra114.c | 5 ++--- arch/arm/mach-tegra/cpuidle-tegra20.c | 4 +--- arch/arm/mach-tegra/cpuidle-tegra30.c | 4 +--- arch/arm/mach-tegra/cpuidle.c | 2 ++ arch/arm/mach-tegra/cpuidle.h | 4 ++++ arch/arm/mach-tegra/pm.c | 5 +++++ arch/arm/mach-tegra/sleep-tegra20.S | 2 +- drivers/gpu/drm/tilcdc/tilcdc_panel.c | 1 - drivers/gpu/drm/tilcdc/tilcdc_slave.c | 1 - drivers/gpu/drm/tilcdc/tilcdc_tfp410.c | 1 - drivers/mfd/ab8500-gpadc.c | 2 +- drivers/misc/Kconfig | 2 +- drivers/mmc/host/sdhci-tegra.c | 2 +- drivers/mtd/nand/Kconfig | 2 +- drivers/spi/spi-sirf.c | 2 +- drivers/staging/comedi/drivers/daqboard2000.c | 14 +++++++------- drivers/staging/nvec/Kconfig | 10 +++++----- drivers/staging/zsmalloc/zsmalloc-main.c | 2 +- drivers/tty/serial/sirfsoc_uart.c | 2 +- drivers/usb/host/ehci-tegra.c | 10 +++++----- drivers/usb/otg/otg.c | 2 +- drivers/usb/phy/tegra_usb_phy.c | 13 +++++++++---- include/linux/mod_devicetable.h | 5 +++-- include/linux/usb/tegra_usb_phy.h | 10 +++++----- sound/soc/ux500/ux500_msp_i2s.h | 1 + 31 files changed, 82 insertions(+), 55 deletions(-) -- 1.8.1.2 Cc: Alan Stern <stern(a)rowland.harvard.edu> Cc: Artem Bityutskiy <artem.bityutskiy(a)linux.intel.com> Cc: Chris Ball <cjb(a)laptop.org> Cc: Dave Airlie <airlied(a)linux.ie> Cc: David Woodhouse <dwmw2(a)infradead.org> Cc: Felipe Balbi <balbi(a)ti.com> Cc: Grant Likely <grant.likely(a)secretlab.ca> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: Mark Brown <broonie(a)kernel.org> Cc: Russell King <linux(a)arm.linux.org.uk> Cc: Samuel Ortiz <sameo(a)linux.intel.com> Cc: Stephen Warren <swarren(a)nvidia.com> Cc: Tony Lindgren <tony(a)atomide.com> Cc: Viresh Kumar <viresh.kumar(a)st.com> Cc: alsa-devel(a)alsa-project.org Cc: dri-devel(a)lists.freedesktop.org Cc: linux-mtd(a)lists.infradead.org Cc: linux-usb(a)vger.kernel.org Cc: spi-devel-general(a)lists.sourceforge.net

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Re: [alsa-devel] BUG: USB audio discontinuities with 'UHCI: implement new semantics for URB_ISO_ASAP'
by Alan Stern 23 Apr '13

23 Apr '13

On Wed, 17 Apr 2013, Joe Rayhawk wrote: > Small buffer/period sizes on usb audio playback though UHCI works fine on v3.7 but causes audio discontinuities/delays on v3.8 and v3.9-rc7. > > I've bisected the behavior down to > > c44b225077bb1fb25ed5cd5c4f226897b91bedd4 > 'UHCI: implement new semantics for URB_ISO_ASAP' > > Tested on two UHCI hosts and three usb-audio devices, all from different vendors. Can you provide a usbmon trace showing the problems? And maybe also a similar trace made under a 3.7 kernel, where the problem doesn't occur? Alan Stern -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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[alsa-devel] [PATCH 2/2] ALSA: ASoc: davinci-mcasp: Add Support BCLK-to-LRCLK ratio for TDM modes
by Michal Bachraty 23 Apr '13

23 Apr '13

For TDM mode, BCLK-to-LCLK ratio is computed as (tdm_slots) x (word_length). I2S mode is only subset of TDM mode with specific tdm_slots = 2 channels. Also bclk_lrclk_ratio can be greater than 255, therefore u16 need to be used. Signed-off-by: Michal Bachraty <michal.bachraty(a)streamunlimited.com> --- sound/soc/davinci/davinci-mcasp.c | 7 ++++--- sound/soc/davinci/davinci-mcasp.h | 2 +- 2 files changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 63633e9..d54e0d9 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -636,11 +636,12 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, * callback, take it into account here. That allows us to for example * send 32 bits per channel to the codec, while only 16 of them carry * audio payload. - * The clock ratio is given for a full period of data (both left and - * right channels), so it has to be divided by 2. + * The clock ratio is given for a full period of data (for I2S format + * both left and right channels), so it has to be divided by number of + * tdm-slots (for I2S - divided by 2). */ if (dev->bclk_lrclk_ratio) - word_length = dev->bclk_lrclk_ratio / 2; + word_length = dev->bclk_lrclk_ratio / dev->tdm_slots; /* mapping of the XSSZ bit-field as described in the datasheet */ fmt = (word_length >> 1) - 1; diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 0edd3b5..a9ac0c1 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -38,7 +38,7 @@ struct davinci_audio_dev { u8 num_serializer; u8 *serial_dir; u8 version; - u8 bclk_lrclk_ratio; + u16 bclk_lrclk_ratio; /* McASP FIFO related */ u8 txnumevt; -- 1.7.9.5

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[alsa-devel] [PATCH 1/2] ALSA: ASoC davinci-pcm, davinci-mcasp Clean up active_serializers
by Michal Bachraty 23 Apr '13

23 Apr '13

As pointed of by Vaibhav, commit message: "ASoC: davinci-mcasp: Add support for multichannel playback" number of active serializers can be hidden into fifo_level variable, which is set in davimci-mcasp. Signed-off-by: Michal Bachraty <michal.bachraty(a)streamunlimited.com> --- sound/soc/davinci/davinci-mcasp.c | 8 +++++--- sound/soc/davinci/davinci-pcm.c | 11 +++++------ sound/soc/davinci/davinci-pcm.h | 1 - 3 files changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 8a9c0ee..63633e9 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -834,17 +834,20 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, int word_length; u8 fifo_level; u8 slots = dev->tdm_slots; + u8 active_serializers; int channels; struct snd_interval *pcm_channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); channels = pcm_channels->min; + active_serializers = (channels + slots - 1) / slots; + if (davinci_hw_common_param(dev, substream->stream, channels) == -EINVAL) return -EINVAL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - fifo_level = dev->txnumevt; + fifo_level = dev->txnumevt * active_serializers; else - fifo_level = dev->rxnumevt; + fifo_level = dev->rxnumevt * active_serializers; if (dev->op_mode == DAVINCI_MCASP_DIT_MODE) davinci_hw_dit_param(dev); @@ -889,7 +892,6 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, dma_params->acnt = dma_params->data_type; dma_params->fifo_level = fifo_level; - dma_params->active_serializers = (channels + slots - 1) / slots; davinci_config_channel_size(dev, word_length); return 0; diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 0161579..8460edc 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -182,7 +182,6 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) unsigned short acnt; unsigned int count; unsigned int fifo_level; - unsigned char serializers = prtd->params->active_serializers; period_size = snd_pcm_lib_period_bytes(substream); dma_offset = prtd->period * period_size; @@ -196,14 +195,14 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) data_type = prtd->params->data_type; count = period_size / data_type; if (fifo_level) - count /= fifo_level * serializers; + count /= fifo_level; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { src = dma_pos; dst = prtd->params->dma_addr; src_bidx = data_type; dst_bidx = 4; - src_cidx = data_type * fifo_level * serializers; + src_cidx = data_type * fifo_level; dst_cidx = 0; } else { src = prtd->params->dma_addr; @@ -211,7 +210,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) src_bidx = 0; dst_bidx = data_type; src_cidx = 0; - dst_cidx = data_type * fifo_level * serializers; + dst_cidx = data_type * fifo_level; } acnt = prtd->params->acnt; @@ -226,8 +225,8 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) ASYNC); else edma_set_transfer_params(prtd->asp_link[0], acnt, - fifo_level * serializers, - count, fifo_level * serializers, + fifo_level, + count, fifo_level, ABSYNC); } diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index 0d84d32..fbb710c 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -27,7 +27,6 @@ struct davinci_pcm_dma_params { unsigned char data_type; /* xfer data type */ unsigned char convert_mono_stereo; unsigned int fifo_level; - unsigned char active_serializers; /* num. of active audio serializers */ }; int davinci_soc_platform_register(struct device *dev); -- 1.7.9.5

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[alsa-devel] [PATCH 1/2] ALSA: ASoc: spdif_transciever: Add DT support
by Michal Bachraty 23 Apr '13

23 Apr '13

Add devicetree support for this dummy audio soc driver. Signed-off-by: Michal Bachraty <michal.bachraty(a)streamunlimited.com> --- sound/soc/codecs/spdif_transciever.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/codecs/spdif_transciever.c b/sound/soc/codecs/spdif_transciever.c index 112a49d..4c8c295 100644 --- a/sound/soc/codecs/spdif_transciever.c +++ b/sound/soc/codecs/spdif_transciever.c @@ -20,6 +20,7 @@ #include <sound/soc.h> #include <sound/pcm.h> #include <sound/initval.h> +#include <linux/of.h> #define DRV_NAME "spdif-dit" @@ -52,12 +53,21 @@ static int spdif_dit_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_OF +static const struct of_device_id spdif_dit_dt_ids[] = { + { .compatible = DRV_NAME, }, + { } +}; +MODULE_DEVICE_TABLE(of, spdif_dits_dt_ids); +#endif + static struct platform_driver spdif_dit_driver = { .probe = spdif_dit_probe, .remove = spdif_dit_remove, .driver = { .name = DRV_NAME, .owner = THIS_MODULE, + .of_match_table = of_match_ptr(spdif_dit_dt_ids), }, }; -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: generic-dmaengine-pcm: call dma_request_slave_channel()
by Shawn Guo 23 Apr '13

23 Apr '13

dma_request_slave_channel() is a more appropriate API for dmaengine clients that adopt generic DMA bindings to call. Let's use it instead of of_dma_request_slave_channel() to save <linux/of_dma.h> include. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- sound/soc/soc-generic-dmaengine-pcm.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 98e1310..e29ec3c 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -21,7 +21,6 @@ #include <sound/soc.h> #include <linux/dma-mapping.h> #include <linux/of.h> -#include <linux/of_dma.h> #include <sound/dmaengine_pcm.h> @@ -219,19 +218,19 @@ static const char * const dmaengine_pcm_dma_channel_names[] = { }; static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, - struct device_node *of_node) + struct device *dev) { unsigned int i; - if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_DT) || !of_node) + if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_DT) || !dev->of_node) return; if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) { - pcm->chan[0] = of_dma_request_slave_channel(of_node, "rx-tx"); + pcm->chan[0] = dma_request_slave_channel(dev, "rx-tx"); pcm->chan[1] = pcm->chan[0]; } else { for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) { - pcm->chan[i] = of_dma_request_slave_channel(of_node, + pcm->chan[i] = dma_request_slave_channel(dev, dmaengine_pcm_dma_channel_names[i]); } } @@ -255,7 +254,7 @@ int snd_dmaengine_pcm_register(struct device *dev, pcm->config = config; pcm->flags = flags; - dmaengine_pcm_request_chan_of(pcm, dev->of_node); + dmaengine_pcm_request_chan_of(pcm, dev); if (flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE) return snd_soc_add_platform(dev, &pcm->platform, -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: generic-dmaengine-pcm: use a more common dma name
by Shawn Guo 23 Apr '13

23 Apr '13

The examples in Documentation/devicetree/bindings/dma/dma.txt recommends the name for dma channel doing both RX and TX to be "rx-tx". This becomes a common pattern that has been adopted by platforms that converts to generic DMA bindings. Let's follow this common pattern in generic-dmaengine-pcm. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- sound/soc/soc-generic-dmaengine-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 5fd5ed4..98e1310 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -227,7 +227,7 @@ static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, return; if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) { - pcm->chan[0] = of_dma_request_slave_channel(of_node, "tx_rx"); + pcm->chan[0] = of_dma_request_slave_channel(of_node, "rx-tx"); pcm->chan[1] = pcm->chan[0]; } else { for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) { -- 1.7.9.5

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[alsa-devel] [PATCH, for-next] ALSA: usb-audio: use FMTBITs in parse_audio_format_i
by Eldad Zack 23 Apr '13

23 Apr '13

Replace the usage of SNDRV_PCM_FORMAT_* macros with the equivalent SNDRV_PCM_FMTBIT_* macros, and the result be can assigned directly to the formats field. Modify the variable name to reflect the change. The following sparse messages are silenced: sound/usb/format.c:377:44: warning: incorrect type in assignment (different base types) sound/usb/format.c:377:44: expected int [signed] pcm_format sound/usb/format.c:377:44: got restricted snd_pcm_format_t [usertype] <noident> sound/usb/format.c:379:44: warning: incorrect type in assignment (different base types) sound/usb/format.c:379:44: expected int [signed] pcm_format sound/usb/format.c:379:44: got restricted snd_pcm_format_t [usertype] <noident> sound/usb/format.c:382:36: warning: incorrect type in assignment (different base types) sound/usb/format.c:382:36: expected int [signed] pcm_format sound/usb/format.c:382:36: got restricted snd_pcm_format_t [usertype] <noident> Signed-off-by: Eldad Zack <eldad(a)fogrefinery.com> --- sound/usb/format.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/usb/format.c b/sound/usb/format.c index 020ede0..e025e28 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -365,7 +365,7 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, { struct usb_interface_descriptor *altsd = get_iface_desc(iface); int protocol = altsd->bInterfaceProtocol; - int pcm_format, ret; + int pcm_formats, ret; if (fmt->bFormatType == UAC_FORMAT_TYPE_III) { /* FIXME: the format type is really IECxxx @@ -377,14 +377,14 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */ if (chip->setup == 0x00 && fp->altsetting == 6) - pcm_format = SNDRV_PCM_FORMAT_S16_BE; + pcm_formats = SNDRV_PCM_FMTBIT_S16_BE; else - pcm_format = SNDRV_PCM_FORMAT_S16_LE; + pcm_formats = SNDRV_PCM_FMTBIT_S16_LE; break; default: - pcm_format = SNDRV_PCM_FORMAT_S16_LE; + pcm_formats = SNDRV_PCM_FMTBIT_S16_LE; } - fp->formats = 1uLL << pcm_format; + fp->formats = pcm_formats; } else { fp->formats = parse_audio_format_i_type(chip, fp, format, fmt, protocol); -- 1.8.1.5

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[alsa-devel] [RFC] ASoC: Current state of Dynamic PCM API
by Sebastian Hesselbarth 22 Apr '13

22 Apr '13

Hi, I am rewriting Marvell Kirkwood ASoC driver to also support Dove and reflect the audio controller's capabilities. The audio controller can have one DMA stream for playback and capture respectively. While capture source can be either I2S or SPDIF, playback can be on both interfaces simultaneously. The setup I have in mind for a generic Kirkwood/Dove soundcard is: (DMA)-->(CPU_DAI)--+<-->(I2S Codec) | +--->(SPDIF TX Codec) | +<---(SPDIF RX Codec) I sent a mail on how to handle multiple codecs on a single cpu_dai earlier, and Mark pointed me to Dynamic PCM API. Unfortunately, current mainline ASoC has no driver using DPCM as an example and integration of DPCM patches look kind-of incomplete to me. Can any of you point me to patches, repositories, or documentation about DPCM where I can start with? Regards, Sebastian

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