June 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/4] ASoC: fsi/rsnd: cyclic transfer support
by Kuninori Morimoto 18 Jun '14

18 Jun '14

Hi Mark Current Renesas sound driver (= FSI/RSND) is not using ALSA common DMAEngine method for some reasons. The 1st reason was that Renesas DMA didn't support cyclic transfer mode. But now, it is supported ! These patches support cyclic transfer for FSI/RSND It needs this branch git://git.infradead.org/users/vkoul/slave-dma.git :: next especially dfbb85cab5f0819d0424a3637b03e7892704fa42 (DMA: shdma: add cyclic transfer support) These are based on mark/topic/rcar branch But, #4 patch needs mark/fix/rcar merge Kuninori Morimoto (4): ASoC: fsi: use SNDRV_DMA_TYPE_DEV for sound buffer ASoC: fsi: add fsi_pointer_update() for common pointer method ASoC: fsi: use dmaengine_prep_dma_cyclic() for DMA transfer ASoC: rsnd: use dmaengine_prep_dma_cyclic() instead of original method arch/arm/mach-shmobile/board-armadillo800eva.c | 4 + arch/arm/mach-shmobile/board-kzm9g.c | 2 + arch/arm/mach-shmobile/board-mackerel.c | 4 + arch/sh/boards/mach-ecovec24/setup.c | 2 + sound/soc/sh/fsi.c | 189 +++++++----------------- sound/soc/sh/rcar/core.c | 76 +++------- sound/soc/sh/rcar/rsnd.h | 4 - 7 files changed, 82 insertions(+), 199 deletions(-) Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH 0/2] ASoC: rsnd: add BUSIF support for DMA
by Kuninori Morimoto 18 Jun '14

18 Jun '14

Hi Mark These patches are for R-Car sound BUSIF support for DMA transfer. If we use BUSIF via DMA, it will work as FIFO. It can reduce underrun error. Kuninori Morimoto (2): ASoC: rsnd: fixup loop exit timing of dma name search ASoC: rsnd: SSI + DMA can select BUSIF .../devicetree/bindings/sound/renesas,rsnd.txt | 1 + include/sound/rcar_snd.h | 1 + sound/soc/sh/rcar/core.c | 26 +++++--- sound/soc/sh/rcar/gen.c | 64 +++++++++++++------- sound/soc/sh/rcar/rsnd.h | 10 ++- sound/soc/sh/rcar/src.c | 37 ++++++++--- sound/soc/sh/rcar/ssi.c | 33 +++++++++- 7 files changed, 126 insertions(+), 46 deletions(-) Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH 0/3] Auvitek au0828 fixups
by Mauro Carvalho Chehab 18 Jun '14

18 Jun '14

Testing auvitek au0828-based devices with xawtv caused lots of overruns, as the period_bytes is set to a too low value, with causes troubles when syncing the audio capture from au0828 card with the audio playback, at the main audio card. It turns that we need a period_bytes size that it is big enough to receive data from two consecutive URB intervals. While here, also add the missing USB IDs from other au0828-based devices. Mauro Carvalho Chehab (3): sound: Add a quirk to enforce period_bytes sound: simplify au0828 quirk table sound: Update au0828 quirks table sound/usb/card.h | 1 + sound/usb/pcm.c | 34 ++++++++++ sound/usb/quirks-table.h | 167 ++++++++++++----------------------------------- sound/usb/quirks.c | 14 ++-- sound/usb/stream.c | 1 + sound/usb/usbaudio.h | 3 +- 6 files changed, 88 insertions(+), 132 deletions(-) -- 1.9.3

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[alsa-devel] Trying to enable digital audio output - Intel HDA - AD1983
by Scott D'Vileskis 18 Jun '14

18 Jun '14

Hi-- I have a Lenovo M55 SFF desktop I use for a media center computer (XBMCbuntu). It has the AD1983 chip on it. From the factory, it had 2 ports in the back (mic + speaker) , and 2 ports in the front (mic + speaker). It also has an on-board speaker. It never shipped with a SPDIF output, but I soldered a wire to the AD1983 and installed an RCA jack on the back of the case. Earlier versions of XBMC shipped with a different version of alsa, and I was able to use this custom SPDIF output without issue for the last year or two, but the current version only enables the 'proper' inputs/outputs I have tried editing the module options and specifying several different models, but the documentation doesn't list any 'model' options for the AD1983. I suspect I need to overwrite the pin configs. cat init_pin_configs: 0x05 0x01014010 0x06 0x0221401f 0x07 0x991301f0 0x08 0x02a19020 0x09 0x0181302e 0x0a 0x414511f0 [ 2421.907510] hda-codec: reconfiguring [ 2421.980324] autoconfig: line_outs=1 (0x5/0x0/0x0/0x0/0x0) type:line [ 2421.980328] speaker_outs=1 (0x7/0x0/0x0/0x0/0x0) [ 2421.980330] hp_outs=1 (0x6/0x0/0x0/0x0/0x0) [ 2421.980332] mono: mono_out=0x0 [ 2421.980333] inputs: [ 2421.980335] Mic=0x8 [ 2421.980336] Line=0x9 [ 2421.981858] input: HDA Intel Front Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card0/input34 [ 2421.984829] input: HDA Intel Line Out as /devices/pci0000:00/0000:00:1b.0/sound/card0/input33 [ 2421.985848] input: HDA Intel Line as /devices/pci0000:00/0000:00:1b.0/sound/card0/input32 Does anyone have any guidance for how I may re-enable the SPDIF on the AD1983 when my BIOS disables it by default? Where else should I look for information on pin configs? Thanks!

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[alsa-devel] [PATCH 1/3] ASoC: Add SND_SOC_BYTES_EXT control type
by Charles Keepax 17 Jun '14

17 Jun '14

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index ed9e2d7..92c1b99 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -256,6 +256,13 @@ ((unsigned long)&(struct soc_bytes) \ {.base = xbase, .num_regs = xregs }) } +#define SND_SOC_BYTES_EXT(xname, xbase, xregs, xhandler_get, xhandler_put) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_bytes_info, .get = xhandler_get, \ + .put = xhandler_put, .private_value = \ + ((unsigned long)&(struct soc_bytes) \ + {.base = xbase, .num_regs = xregs }) } + #define SND_SOC_BYTES_MASK(xname, xbase, xregs, xmask) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_bytes_info, .get = snd_soc_bytes_get, \ -- 1.7.2.5

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[alsa-devel] alsa-lib: pcm_plug: fix float conversion for user specified ttable
by Sergey 17 Jun '14

17 Jun '14

Move custom ttable with equal channels case from a separate ttable_last exception into a common plugins insertion loop. Fixes plug with ttable for float pcms (jack, ladspa). Example: aplay -fFLOAT_LE /dev/zero pcm.!default { type plug slave.pcm { type null } ttable.0.0 1 } --- src/pcm/pcm_plug.c | 36 +++++++++++------------------------- 1 file changed, 11 insertions(+), 25 deletions(-) diff --git a/src/pcm/pcm_plug.c b/src/pcm/pcm_plug.c index ede9c15..6d0454f 100644 --- a/src/pcm/pcm_plug.c +++ b/src/pcm/pcm_plug.c @@ -53,7 +53,7 @@ typedef struct { const snd_config_t *rate_converter; enum snd_pcm_plug_route_policy route_policy; snd_pcm_route_ttable_entry_t *ttable; - int ttable_ok, ttable_last; + int ttable_ok; unsigned int tt_ssize, tt_cused, tt_sused; } snd_pcm_plug_t; @@ -380,7 +380,7 @@ static int snd_pcm_plug_change_channels(snd_pcm_t *pcm, snd_pcm_t **new, snd_pcm snd_pcm_route_ttable_entry_t *ttable; int err; if (clt->channels == slv->channels && - (!plug->ttable || !plug->ttable_last)) + (!plug->ttable || plug->ttable_ok)) return 0; if (clt->rate != slv->rate && clt->channels > slv->channels) @@ -485,13 +485,15 @@ static int snd_pcm_plug_change_format(snd_pcm_t *pcm, snd_pcm_t **new, snd_pcm_p /* No conversion is needed */ if (clt->format == slv->format && clt->rate == slv->rate && - clt->channels == slv->channels) + clt->channels == slv->channels && + (!plug->ttable || plug->ttable_ok)) return 0; if (snd_pcm_format_linear(slv->format)) { /* Conversion is done in another plugin */ if (clt->rate != slv->rate || - clt->channels != slv->channels) + clt->channels != slv->channels || + (plug->ttable && !plug->ttable_ok)) return 0; cfmt = clt->format; switch (clt->format) { @@ -525,7 +527,8 @@ static int snd_pcm_plug_change_format(snd_pcm_t *pcm, snd_pcm_t **new, snd_pcm_p if (snd_pcm_format_linear(clt->format)) { cfmt = clt->format; f = snd_pcm_lfloat_open; - } else if (clt->rate != slv->rate || clt->channels != slv->channels) { + } else if (clt->rate != slv->rate || clt->channels != slv->channels || + (plug->ttable && !plug->ttable_ok)) { cfmt = SND_PCM_FORMAT_S16; f = snd_pcm_lfloat_open; } else @@ -641,11 +644,12 @@ static int snd_pcm_plug_insert_plugins(snd_pcm_t *pcm, }; snd_pcm_plug_params_t p = *slave; unsigned int k = 0; - plug->ttable_ok = plug->ttable_last = 0; + plug->ttable_ok = 0; while (client->format != p.format || client->channels != p.channels || client->rate != p.rate || - client->access != p.access) { + client->access != p.access || + (plug->ttable && !plug->ttable_ok)) { snd_pcm_t *new; int err; if (k >= sizeof(funcs)/sizeof(*funcs)) @@ -660,24 +664,6 @@ static int snd_pcm_plug_insert_plugins(snd_pcm_t *pcm, } k++; } -#ifdef BUILD_PCM_PLUGIN_ROUTE - /* it's exception, user specified ttable, but no reduction/expand */ - if (plug->ttable && !plug->ttable_ok) { - snd_pcm_t *new; - int err; - plug->ttable_last = 1; - err = snd_pcm_plug_change_channels(pcm, &new, client, &p); - if (err < 0) { - snd_pcm_plug_clear(pcm); - return err; - } - assert(err); - assert(plug->ttable_ok); - plug->gen.slave = new; - pcm->fast_ops = new->fast_ops; - pcm->fast_op_arg = new->fast_op_arg; - } -#endif return 0; } -- 1.7.10.4

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[alsa-devel] [PATCH v3 0/3] Add ES8328 audio codec
by Sean Cross 17 Jun '14

17 Jun '14

This patchset adds support for the Everest Semi ES8328 audio codec. It also adds support for using the es8328 on IMX boards. We write a machine driver rather than using simple-card because the machine driver needs to support regulators for the codec itself and speaker amps, have headphone support, and must supply an unusual frequency to the codec chip. Changes since v2: - Change deemph to a bool, and update based on sample rate - Replace value enums with regular enums - Break SPI and I2C drivers into their own modules - Rename everest,es8328 to es8328 as the module wasn't getting detected. Because of this, the patch to vendor-prefixes.txt has been dropped. Changes since v1: - Rename HP/Speaker to OUT1/OUT2 - Use DAPM widgets for startup/shutdown - Add regulator support to machine driver *** BLURB HERE *** Sean Cross (2): ASoC: add es8328 codec driver ASoC: fsl: add imx-es8328 machine driver Documentation/devicetree/bindings/sound/es8328.txt | 18 + .../devicetree/bindings/sound/imx-audio-es8328.txt | 60 ++ sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/es8328-i2c.c | 55 ++ sound/soc/codecs/es8328-spi.c | 44 ++ sound/soc/codecs/es8328.c | 625 +++++++++++++++++++++ sound/soc/codecs/es8328.h | 313 +++++++++++ sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-es8328.c | 359 ++++++++++++ 11 files changed, 1509 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/es8328.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-es8328.txt create mode 100644 sound/soc/codecs/es8328-i2c.c create mode 100644 sound/soc/codecs/es8328-spi.c create mode 100644 sound/soc/codecs/es8328.c create mode 100644 sound/soc/codecs/es8328.h create mode 100644 sound/soc/fsl/imx-es8328.c -- 2.0.0

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[alsa-devel] ASoC updates for v3.16
by Mark Brown 17 Jun '14

17 Jun '14

The following changes since commit 7171511eaec5bf23fb06078f59784a3a0626b38f: Linux 3.16-rc1 (2014-06-15 17:45:28 -1000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.16-rc1 for you to fetch changes up to 28e48f0e262fd94a613613d06ac5eaf6745b18a8: Merge remote-tracking branches 'asoc/fix/fsl-dma', 'asoc/fix/fsl-spdif', 'asoc/fix/pxa', 'asoc/fix/rcar' and 'asoc/fix/sigmadsp' into asoc-linus (2014-06-16 16:05:16 +0100) ---------------------------------------------------------------- ASoC: Fixes for v3.16 Quite a few build coverage fixes in here among the usual small driver fixes includling the sigmadsp change from Lars - moving the driver to separate modules per bus (which is basically just code motion) avoids issues with some combinations of buses being enabled. ---------------------------------------------------------------- Anssi Hannula (1): ASoC: fsl_spdif: Fix integer overflow when calculating divisors Arnd Bergmann (2): ASoC: pxa: add I2C dependencies as needed ASoC: MMP audio needs sram support Guenter Roeck (1): ASoC: fsl: Fix build problem Jarkko Nikula (1): ASoC: dapm: Make sure register value is in sync with DAPM kcontrol state Kuninori Morimoto (1): ASoC: rsnd: fixup index of src/dst mod when capture Lars-Peter Clausen (1): ASoC: sigmadsp: Split regmap and I2C support into separate modules Mark Brown (2): Merge remote-tracking branch 'asoc/fix/core' into asoc-linus Merge remote-tracking branches 'asoc/fix/fsl-dma', 'asoc/fix/fsl-spdif', 'asoc/fix/pxa', 'asoc/fix/rcar' and 'asoc/fix/sigmadsp' into asoc-linus Nicolin Chen (1): ASoC: fsl_spdif: Fix incorrect usage of regmap_read() sound/soc/codecs/Kconfig | 12 +++++-- sound/soc/codecs/Makefile | 4 +++ sound/soc/codecs/sigmadsp-i2c.c | 35 ++++++++++++++++++++ sound/soc/codecs/sigmadsp-regmap.c | 36 +++++++++++++++++++++ sound/soc/codecs/sigmadsp.c | 65 ++------------------------------------ sound/soc/codecs/sigmadsp.h | 20 ++++++++++++ sound/soc/fsl/fsl_dma.c | 4 +-- sound/soc/fsl/fsl_spdif.c | 6 ++-- sound/soc/pxa/Kconfig | 11 ++++--- sound/soc/sh/rcar/core.c | 2 +- sound/soc/soc-dapm.c | 29 +++++++++-------- 11 files changed, 135 insertions(+), 89 deletions(-) create mode 100644 sound/soc/codecs/sigmadsp-i2c.c create mode 100644 sound/soc/codecs/sigmadsp-regmap.c

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[alsa-devel] TLV320AIC32x4 driver configuration for playing wav file using aplay
by Rahul Panwar 17 Jun '14

17 Jun '14

Hi, We integrated TLV320AIC32x4 codec to OMAP-L138 Soc board. We are using Linux kernel 3.3.0 with driver of TLV320AIC32x4. Driver is successfully detecting. But we are not able to play a wav file using aplay. When we are trying to play any WAV file using aplay, it is not enabling the DAC and file failed to play. When we try to debug that using debugfs, it look like, it is only setting "LOR" ==> "LOR Powe" ==> "LOR Output Mixer". But after that there is no further Input / path / output configuration found. As per the driver it should enable the "Right DAC" via "R_DAC Switch". When it will enable "Right DAC", DAC will be enabled and it may start playing WAV file after that. We are not able the get the cause, what we are doing wrong. What should we need to do to enable DAC. I am attaching the debugfs log for TLV320AIC32x4 dapm, it may be helpful to understand the cause, which we are unable to get. -- Thanks and Regards, Rahul Panwar www.linuxexplore.com

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[alsa-devel] [PATCH] ASoC: wm8994: Prevent double lock of accdet_lock mutex on wm1811
by Charles Keepax 17 Jun '14

17 Jun '14

wm1811_micd_stop takes the accdet_lock mutex, and is called from two places, one of which is already holding the accdet_lock. This obviously causes a lock up. This patch fixes this issue by removing the lock from wm1811_micd_stop and ensuring that it is always locked externally. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 247b390..9719d3c 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3505,6 +3505,7 @@ static irqreturn_t wm8994_mic_irq(int irq, void *data) return IRQ_HANDLED; } +/* Should be called with accdet_lock held */ static void wm1811_micd_stop(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); @@ -3512,14 +3513,10 @@ static void wm1811_micd_stop(struct snd_soc_codec *codec) if (!wm8994->jackdet) return; - mutex_lock(&wm8994->accdet_lock); - snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, WM8958_MICD_ENA, 0); wm1811_jackdet_set_mode(codec, WM1811_JACKDET_MODE_JACK); - mutex_unlock(&wm8994->accdet_lock); - if (wm8994->wm8994->pdata.jd_ext_cap) snd_soc_dapm_disable_pin(&codec->dapm, "MICBIAS2"); @@ -3560,10 +3557,10 @@ static void wm8958_open_circuit_work(struct work_struct *work) open_circuit_work.work); struct device *dev = wm8994->wm8994->dev; - wm1811_micd_stop(wm8994->hubs.codec); - mutex_lock(&wm8994->accdet_lock); + wm1811_micd_stop(wm8994->hubs.codec); + dev_dbg(dev, "Reporting open circuit\n"); wm8994->jack_mic = false; -- 1.7.9.5

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