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

[alsa-devel] [PATCH 0/3] Yamaha THR10 and related USB quirks
by Trulan Martin 24 Apr '13

24 Apr '13

The following patches add USB quirks for several Yamaha amps. Please CC me as I am not subscribed to the mailing list.

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[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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