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

[alsa-devel] [PATCH] ASoC: Remove name_prefix unset during DAI link init hack again
by Lars-Peter Clausen 26 Apr '14

26 Apr '14

This was initially removed in commit 6479f15ad ("ASoC: Remove name_prefix unset during DAI link init hack"), but was brought back in commit 503ae5e0 ("ASoC: core: Add helpers for dai link and aux dev init") by accident. This patch removes it again. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-core.c | 13 ------------- 1 file changed, 13 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f18112a..94c1668 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1272,14 +1272,10 @@ static int soc_aux_dev_init(struct snd_soc_card *card, { struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; - const char *temp; int ret; rtd->card = card; - temp = codec->name_prefix; - codec->name_prefix = NULL; - /* do machine specific initialization */ if (aux_dev->init) { ret = aux_dev->init(&codec->dapm); @@ -1287,8 +1283,6 @@ static int soc_aux_dev_init(struct snd_soc_card *card, return ret; } - codec->name_prefix = temp; - rtd->codec = codec; return 0; @@ -1300,15 +1294,10 @@ static int soc_dai_link_init(struct snd_soc_card *card, { struct snd_soc_dai_link *dai_link = &card->dai_link[num]; struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; - const char *temp; int ret; rtd->card = card; - /* machine controls, routes and widgets are not prefixed */ - temp = codec->name_prefix; - codec->name_prefix = NULL; - /* do machine specific initialization */ if (dai_link->init) { ret = dai_link->init(rtd); @@ -1316,8 +1305,6 @@ static int soc_dai_link_init(struct snd_soc_card *card, return ret; } - codec->name_prefix = temp; - rtd->codec = codec; return 0; -- 1.8.0

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[alsa-devel] [PATCH] ASoC: fsl-ssi: Allow to use driver with generic audio card
by Alexander Shiyan 26 Apr '14

26 Apr '14

Generic sound card checks return value of functions for setup format, clock and slot for -ENOTSUPP error. Add this compatibility to fsl-ssi driver. --- I'm really not sure that's a good solution, please correct me if I'm wrong. Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- sound/soc/fsl/fsl_ssi.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 25865d8f..6891b8c 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -764,7 +764,7 @@ static int fsl_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) ssi_private->i2s_mode |= CCSR_SSI_SCR_I2S_MODE_SLAVE; break; default: - return -EINVAL; + return -ENOTSUPP; } /* Data on rising edge of bclk, frame low, 1clk before data */ @@ -789,7 +789,7 @@ static int fsl_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) ssi_private->i2s_mode |= CCSR_SSI_SCR_I2S_MODE_NORMAL; break; default: - return -EINVAL; + return -ENOTSUPP; } scr |= ssi_private->i2s_mode; @@ -812,7 +812,7 @@ static int fsl_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) strcr ^= CCSR_SSI_STCR_TFSI; break; default: - return -EINVAL; + return -ENOTSUPP; } /* DAI clock master masks */ @@ -825,7 +825,7 @@ static int fsl_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) scr &= ~CCSR_SSI_SCR_SYS_CLK_EN; break; default: - return -EINVAL; + return -ENOTSUPP; } stcr |= strcr; @@ -897,7 +897,7 @@ static int fsl_ssi_set_dai_sysclk(struct snd_soc_dai *cpu_dai, /* Don't apply it to any non-baudclk circumstance */ if (IS_ERR(ssi_private->baudclk)) - return -EINVAL; + return -ENOTSUPP; /* It should be already enough to divide clock by setting pm alone */ psr = 0; @@ -986,7 +986,7 @@ static int fsl_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask, val = read_ssi(&ssi->scr) & (CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_NET); if (val && slots < 2) { dev_err(cpu_dai->dev, "slot number should be >= 2 in I2S or NET\n"); - return -EINVAL; + return -ENOTSUPP; } write_ssi_mask(&ssi->stccr, CCSR_SSI_SxCCR_DC_MASK, -- 1.8.3.2

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[alsa-devel] [PATCH RFC 00/17] Rework OMAP4+ HDMI audio support
by Jyri Sarha 26 Apr '14

26 Apr '14

This set of patches fixes OMAP4+ HDMI audio. The structure of the implementatin looks a bit different than before. Instead of creating a driver specific API for a separate ASoC component driver to connect to, this implementation integrates an the component driver into the HDMI driver. The idea is to use an existing ASoC component driver API instead of creating a new custom API for each HDMI IP and to avoid splitting the driver to half for separate video and audio parts connected with the API. The new implementation also uses simple-audio-card for a machine driver instead of having its own HW specific machine driver. The patches are based on 3.15-rc2 merged with git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next and git://git.kernel.org/pub/scm/linux/kernel/git/tomba/linux.git work/dss-dt-omap5 Everything is pushed here here: git://git.ti.com/~jyrisarha/ti-linux-kernel/jyrisarhas-audio-video-linux-fe… omap-hdmi-audio Best regards, Jyri Jyri Sarha (17): ASoC: omap-pcm: Move omap-pcm under include/sound OMAPDSS: hdmi-common: Add hdmi_dss_audio_from_hw_params() OMAPDSS: hdmi4: Integrated ASoC DAI component driver implementation OMAPDSS: Kconfig: Add depencies and help section to OMAP4_DSS_HDMI_AUDIO OMAPDSS: hdmi.h: Add HDMI_AUDIO_LAYOUT_6CH enum value OMAPDSS: hdmi5_core: Fix compilation with OMAP5_DSS_HDMI_AUDIO OMAPDSS: hdmi5: Integrated ASoC DAI component driver implementation OMAPDSS: Kconfig: Add depencies and help section to OMAP5_DSS_HDMI_AUDIO ASoC: omap: Remove obsolete HDMI audio code and Kconfig options OMAPDSS: Remove obsolete audio code ARM: omap4.dtsi: Add audio related parametes to hdmi node ARM: omap4-panda-common.dtsi: Add HDMI audio nodes ARM: omap5.dtsi: Add audio related parameters to hdmi node ARM: omap5-uevm.dts: Add hdmi audio related nodes ARM: omap2plus_defconfig: Build DSS HDMI support for OMAP4 in kernel ARM: omap2plus_defconfig: Enable OMAP5 HDMI support ARM: omap2plus_defconfig: Enable OMAP4+ HDMI audio support .../devicetree/bindings/video/ti,omap4-dss.txt | 4 + arch/arm/boot/dts/omap4-panda-common.dtsi | 21 +- arch/arm/boot/dts/omap4.dtsi | 2 + arch/arm/boot/dts/omap5-uevm.dts | 19 + arch/arm/boot/dts/omap5.dtsi | 2 + arch/arm/configs/omap2plus_defconfig | 20 +- .../fbdev/omap2/displays-new/connector-hdmi.c | 99 ------ .../fbdev/omap2/displays-new/encoder-tpd12s015.c | 56 --- drivers/video/fbdev/omap2/dss/Kconfig | 19 +- drivers/video/fbdev/omap2/dss/hdmi.h | 8 +- drivers/video/fbdev/omap2/dss/hdmi4.c | 216 ++++++++---- drivers/video/fbdev/omap2/dss/hdmi5.c | 216 ++++++++---- drivers/video/fbdev/omap2/dss/hdmi5_core.c | 2 +- drivers/video/fbdev/omap2/dss/hdmi_common.c | 117 ++++++- include/sound/omap-pcm.h | 30 ++ include/video/omapdss.h | 31 -- sound/soc/davinci/davinci-mcasp.c | 2 +- sound/soc/omap/Kconfig | 13 - sound/soc/omap/Makefile | 4 - sound/soc/omap/omap-dmic.c | 2 +- sound/soc/omap/omap-hdmi-card.c | 87 ----- sound/soc/omap/omap-hdmi.c | 364 -------------------- sound/soc/omap/omap-hdmi.h | 38 -- sound/soc/omap/omap-mcbsp.c | 2 +- sound/soc/omap/omap-mcpdm.c | 2 +- sound/soc/omap/omap-pcm.h | 30 -- 26 files changed, 536 insertions(+), 870 deletions(-) create mode 100644 include/sound/omap-pcm.h delete mode 100644 sound/soc/omap/omap-hdmi-card.c delete mode 100644 sound/soc/omap/omap-hdmi.c delete mode 100644 sound/soc/omap/omap-hdmi.h delete mode 100644 sound/soc/omap/omap-pcm.h -- 1.7.9.5

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[alsa-devel] [PATCH 1/4] ASoC: tlv320dac3: Use devm_regulator_bulk_get()
by Fabio Estevam 25 Apr '14

25 Apr '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> Using devm_regulator_bulk_get() can make the code cleaner and smaller as we do not need to call regulator_bulk_free() in the error and remove paths. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/tlv320dac33.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 517055a..df3a750 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1540,7 +1540,7 @@ static int dac33_i2c_probe(struct i2c_client *client, for (i = 0; i < ARRAY_SIZE(dac33->supplies); i++) dac33->supplies[i].supply = dac33_supply_names[i]; - ret = regulator_bulk_get(&client->dev, ARRAY_SIZE(dac33->supplies), + ret = devm_regulator_bulk_get(&client->dev, ARRAY_SIZE(dac33->supplies), dac33->supplies); if (ret != 0) { @@ -1551,11 +1551,9 @@ static int dac33_i2c_probe(struct i2c_client *client, ret = snd_soc_register_codec(&client->dev, &soc_codec_dev_tlv320dac33, &dac33_dai, 1); if (ret < 0) - goto err_register; + goto err_get; return ret; -err_register: - regulator_bulk_free(ARRAY_SIZE(dac33->supplies), dac33->supplies); err_get: if (dac33->power_gpio >= 0) gpio_free(dac33->power_gpio); @@ -1573,8 +1571,6 @@ static int dac33_i2c_remove(struct i2c_client *client) if (dac33->power_gpio >= 0) gpio_free(dac33->power_gpio); - regulator_bulk_free(ARRAY_SIZE(dac33->supplies), dac33->supplies); - snd_soc_unregister_codec(&client->dev); return 0; } -- 1.8.3.2

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[alsa-devel] [PATCH 0/5] treewide: Convert logging message misuses of /n to \n
by Joe Perches 25 Apr '14

25 Apr '14

Joe Perches (5): qlge: Convert /n to \n iwlegacy: Convert /n to \n staging: rtl<foo>: Convert /n to \n appledisplay: Convert /n to \n sound: tlv320aic31xx: Convert /n to \n drivers/net/ethernet/qlogic/qlge/qlge_main.c | 2 +- drivers/net/wireless/iwlegacy/3945.c | 2 +- drivers/net/wireless/iwlegacy/4965-mac.c | 2 +- drivers/staging/rtl8188eu/hal/usb_ops_linux.c | 2 +- drivers/staging/rtl8188eu/os_dep/usb_intf.c | 2 +- drivers/staging/rtl8723au/hal/usb_ops_linux.c | 4 ++-- drivers/staging/rtl8723au/os_dep/usb_intf.c | 2 +- drivers/usb/misc/appledisplay.c | 2 +- sound/soc/codecs/tlv320aic31xx.c | 2 +- 9 files changed, 10 insertions(+), 10 deletions(-) -- 1.8.1.2.459.gbcd45b4.dirty

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[alsa-devel] [PATCH] ASoC: sgtl5000: Use devm_regulator_bulk_get()
by Fabio Estevam 24 Apr '14

24 Apr '14

Using devm_regulator_bulk_get() can make the code cleaner and smaller as we do not need to call regulator_bulk_free() in the error and remove paths. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 11 ++--------- 1 file changed, 2 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index b56caef..9626ee0 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1322,7 +1322,7 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) return ret; } - ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies), + ret = devm_regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); if (ret) goto err_ldo_remove; @@ -1330,16 +1330,13 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); if (ret) - goto err_regulator_free; + goto err_ldo_remove; /* wait for all power rails bring up */ udelay(10); return 0; -err_regulator_free: - regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); err_ldo_remove: if (!external_vddd) ldo_regulator_remove(codec); @@ -1409,8 +1406,6 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) err: regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); - regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); ldo_regulator_remove(codec); return ret; @@ -1424,8 +1419,6 @@ static int sgtl5000_remove(struct snd_soc_codec *codec) regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); - regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); ldo_regulator_remove(codec); return 0; -- 1.8.3.2

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[alsa-devel] [PATCH] ASoC: simple-card: is_top_level_node parameter to simple_card_dai_link_of()
by Jyri Sarha 24 Apr '14

24 Apr '14

Restore correct parsing of dai-link subnodes with more explicit implementation for applying the "simple-audio-card,"-prefix to dai-link property and subnode names. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- sound/soc/generic/simple-card.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 98f97e5..06fe0e2 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -144,7 +144,8 @@ asoc_simple_card_sub_parse_of(struct device_node *np, static int simple_card_dai_link_of(struct device_node *node, struct device *dev, struct snd_soc_dai_link *dai_link, - struct simple_dai_props *dai_props) + struct simple_dai_props *dai_props, + bool is_top_level_node) { struct device_node *np = NULL; struct device_node *bitclkmaster = NULL; @@ -155,7 +156,8 @@ static int simple_card_dai_link_of(struct device_node *node, char *prefix = ""; int ret; - prefix = "simple-audio-card,"; + if (is_top_level_node) + prefix = "simple-audio-card,"; daifmt = snd_soc_of_parse_daifmt(node, prefix, &bitclkmaster, &framemaster); @@ -307,14 +309,15 @@ static int asoc_simple_card_parse_of(struct device_node *node, for (i = 0; (np = of_get_next_child(node, np)); i++) { dev_dbg(dev, "\tlink %d:\n", i); ret = simple_card_dai_link_of(np, dev, dai_link + i, - dai_props + i); + dai_props + i, false); if (ret < 0) { of_node_put(np); return ret; } } } else { - ret = simple_card_dai_link_of(node, dev, dai_link, dai_props); + ret = simple_card_dai_link_of(node, dev, dai_link, dai_props, + true); if (ret < 0) return ret; } -- 1.7.9.5

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Re: [alsa-devel] [PATCH] ASoC: imx-sgtl5000.c: Add support for specifying PDCR_TXRXEN
by Markus Pargmann 24 Apr '14

24 Apr '14

Hi, On Wed, Apr 23, 2014 at 05:50:18PM +0200, David Jander wrote: > This enables effectively swapping RX/TX pins on the external audio port > > Signed-off-by: David Jander <david(a)protonic.nl> > --- > Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt | 4 ++++ > sound/soc/fsl/imx-sgtl5000.c | 5 ++++- > 2 files changed, 8 insertions(+), 1 deletion(-) > > diff --git a/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt b/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt > index e4acdd8..3afea85 100644 > --- a/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt > +++ b/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt > @@ -29,6 +29,10 @@ Required properties: > - mux-int-port : The internal port of the i.MX audio muxer (AUDMUX) > - mux-ext-port : The external port of the i.MX audio muxer > > +Optional properties: > +- ext-port-txrxen : Flag. Set to swap RX/TX signals on external port (set the > + IMX_AUDMUX_V2_PDCR_TXRXEN bit in the PDC register of the external port) > + > Note: The AUDMUX port numbering should start at 1, which is consistent with > hardware manual. You could also try to use the simple sound card and use DT properties to setup audmux. This is a configuration example for the audmux: &audmux { status = "okay"; aud4 { fsl,audmux-port = <MX31_AUDMUX_PORT4_SSI_PINS_4>; fsl,port-config = < (IMX_AUDMUX_V2_PTCR_SYN | IMX_AUDMUX_V2_PTCR_TFSDIR | IMX_AUDMUX_V2_PTCR_TFSEL(MX31_AUDMUX_PORT1_SSI0) | IMX_AUDMUX_V2_PTCR_TCLKDIR | IMX_AUDMUX_V2_PTCR_TCSEL(MX31_AUDMUX_PORT1_SSI0)) IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT1_SSI0) >; }; }; Regards, Markus > > diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c > index 1cb22dd..b543fa6 100644 > --- a/sound/soc/fsl/imx-sgtl5000.c > +++ b/sound/soc/fsl/imx-sgtl5000.c > @@ -63,6 +63,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) > struct i2c_client *codec_dev; > struct imx_sgtl5000_data *data = NULL; > int int_port, ext_port; > + unsigned int ext_txrxen = 0; > int ret; > > ret = of_property_read_u32(np, "mux-int-port", &int_port); > @@ -75,6 +76,8 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) > dev_err(&pdev->dev, "mux-ext-port missing or invalid\n"); > return ret; > } > + if (of_property_read_bool(np, "ext-port-txrxen")) > + ext_txrxen = IMX_AUDMUX_V2_PDCR_TXRXEN; > > /* > * The port numbering in the hardware manual starts at 1, while > @@ -95,7 +98,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) > } > ret = imx_audmux_v2_configure_port(ext_port, > IMX_AUDMUX_V2_PTCR_SYN, > - IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); > + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port) | ext_txrxen); > if (ret) { > dev_err(&pdev->dev, "audmux external port setup failed\n"); > return ret; > -- > 1.9.1 > > -- Pengutronix e.K. | | Industrial Linux Solutions | http://www.pengutronix.de/ | Peiner Str. 6-8, 31137 Hildesheim, Germany | Phone: +49-5121-206917-0 | Amtsgericht Hildesheim, HRA 2686 | Fax: +49-5121-206917-5555 |

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[alsa-devel] [PATCH 0/3] Fix some minor issues in simple card driver
by Nicolin Chen 24 Apr '14

24 Apr '14

The series fixes one name limitation and code style problems. Nicolin Chen (3): ASoC: simple-card: Drop node->name checking ASoC: simple-card: Simplify error msg in simple_card_dai_link_of() ASoC: simple-card: Improve coding style sound/soc/generic/simple-card.c | 32 ++++++++++++++------------------ 1 file changed, 14 insertions(+), 18 deletions(-) -- 1.8.4

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[alsa-devel] [PATCH] ASoC: fsl: Drop formats limitation for imx-pcm-dma.c
by Nicolin Chen 24 Apr '14

24 Apr '14

Now ASoC core is getting the intersection of supported formats not only from CPU and CODEC dai's but also from DMA's. However, there should be no specific width limitation from SDMA side. So drop it. Otherwise, we would only support S16_LE format for all i.MX platforms. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/fsl/imx-pcm-dma.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c index 2585ae4..0849b7b 100644 --- a/sound/soc/fsl/imx-pcm-dma.c +++ b/sound/soc/fsl/imx-pcm-dma.c @@ -40,7 +40,6 @@ static const struct snd_pcm_hardware imx_pcm_hardware = { SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, - .formats = SNDRV_PCM_FMTBIT_S16_LE, .buffer_bytes_max = IMX_SSI_DMABUF_SIZE, .period_bytes_min = 128, .period_bytes_max = 65535, /* Limited by SDMA engine */ -- 1.8.4

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