August 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH V3] ASoC: Davinci: machine: Add device tree binding
by Hebbar, Gururaja 12 Sep '12

12 Sep '12

Device tree support for Davinci Machine driver When the board boots with device tree, the driver will receive card, codec, dai interface details (like the card name, DAPM routing map, phandle for the audio components described in the dts file, codec mclk speed). The card will be set up based on this information. Since the routing is provided via DT we can mark the card fully routed so core can take care of disconnecting the unused pins. When here, code indentation and comment style is also fixed Signed-off-by: Hebbar, Gururaja <gururaja.hebbar(a)ti.com> --- Changes from V1: - Change DT parameter from "ti,codec-clock" to "ti,codec-clock-rate" - add more explanation to DT parameters Changes from V2: - Update Documentation details - Remove irrelevant commit message :000000 100644 0000000... b248014... A Documentation/devicetree/bindings/sound/davinci-evm-audio.txt :100644 100644 ab0ad45... ca2a547... M sound/soc/davinci/davinci-evm.c .../bindings/sound/davinci-evm-audio.txt | 66 +++++++ sound/soc/davinci/davinci-evm.c | 185 +++++++++++++++++-- 2 files changed, 231 insertions(+), 20 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt b/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt new file mode 100644 index 0000000..b248014 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt @@ -0,0 +1,66 @@ +* Texas Instruments SoC audio setups with TLV320AIC3X Codec + +Required properties: +- compatible : "ti,davinci-evm-audio" +- ti,model : The user-visible name of this sound complex. +- ti,audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the codec's pins, and the jacks on the board: + + Codec pins: + + * MIC3L + * MIC3R + * LINE1L + * LINE2L + * LINE1R + * LINE2R + + Board connectors: + + * Headphone Jack + * Line Out + * Mic Jack + +- ti,mcasp-controller : The phandle of the McASP controller +- ti,audio-codec : The phandle of the TLV320AIC3x audio codec +- ti,codec-clock-rate : The Codec Clock rate (in Hz) applied to the Codec + +- dai-data : A list of DAI data used by SOC code to register + DAI, Codecs platform. + The string index "should" be as shown below. +dai-data = +"<DAI Name>", "<DAI Stream Name>", +"<CODEC DAI Name>", "<true if evm_aic3x_init is required, else false>", +"<evm ops required (evm_ops or evm_spdif_ops)>"; + +Here fields +"<DAI Name>" : used to indicate the DAI Name +"<DAI Stream Name>" : used to indicate the Stream Name +"<CODEC DAI Name>" : used to bind the link between Codec DAI and ASOC DAI + +Machine related options +"<true/false>" : Whether the machine specific initialization + : evm_aic3x_init() is required + +"<evm ops required>" : Which hardware ops function is to be used. + : (evm_ops or evm_spdif_ops) + : use evm-spdif-ops if DAI is working in DIT mode + : else use evm-ops. These ops setup hw param callbacks + : which are used to setup CODEC/cpu DAI configuration + : and codec system clock. + +Example: + +sound { + compatible = "ti,davinci-evm-audio"; + ti,model = "DA830 EVM"; + ti,audio-codec = <&tlv320aic3x>; + ti,mcasp-controller = <&mcasp1>; + ti,codec-clock-rate = <12000000>; + dai-data = + "TLV320AIC3X", "AIC3X", + "tlv320aic3x-hifi", "true", + "evm-ops"; +}; diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index ab0ad45..ca2a547 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -34,27 +34,38 @@ static int evm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_card *soc_card = codec->card; + struct device_node *np = soc_card->dev->of_node; int ret = 0; unsigned sysclk; - /* ASP1 on DM355 EVM is clocked by an external oscillator */ - if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm() || - machine_is_davinci_dm365_evm()) - sysclk = 27000000; - - /* ASP0 in DM6446 EVM is clocked by U55, as configured by - * board-dm644x-evm.c using GPIOs from U18. There are six - * options; here we "know" we use a 48 KHz sample rate. - */ - else if (machine_is_davinci_evm()) - sysclk = 12288000; - - else if (machine_is_davinci_da830_evm() || - machine_is_davinci_da850_evm()) - sysclk = 24576000; - - else - return -EINVAL; + if (np) { + ret = of_property_read_u32(np, "ti,codec-clock-rate", &sysclk); + if (ret < 0) + return ret; + } else { + /* ASP1 on DM355 EVM is clocked by an external oscillator */ + if (machine_is_davinci_dm355_evm() || + machine_is_davinci_dm6467_evm() || + machine_is_davinci_dm365_evm()) + sysclk = 27000000; + + /* + * ASP0 in DM6446 EVM is clocked by U55, as configured by + * board-dm644x-evm.c using GPIOs from U18. There are six + * options; here we "know" we use a 48 KHz sample rate. + */ + else if (machine_is_davinci_evm()) + sysclk = 12288000; + + else if (machine_is_davinci_da830_evm() || + machine_is_davinci_da850_evm()) + sysclk = 24576000; + + else + return -EINVAL; + } /* set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, AUDIO_FORMAT); @@ -127,13 +138,22 @@ static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; + struct device_node *np = codec->card->dev->of_node; + int ret; /* Add davinci-evm specific widgets */ snd_soc_dapm_new_controls(dapm, aic3x_dapm_widgets, ARRAY_SIZE(aic3x_dapm_widgets)); - /* Set up davinci-evm specific audio path audio_map */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); + if (np) { + ret = snd_soc_of_parse_audio_routing(codec->card, + "ti,audio-routing"); + if (ret) + return ret; + } else { + /* Set up davinci-evm specific audio path audio_map */ + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); + } /* not connected */ snd_soc_dapm_disable_pin(dapm, "MONO_LOUT"); @@ -282,6 +302,113 @@ static struct snd_soc_card da850_snd_soc_card = { .num_links = 1, }; +#if defined(CONFIG_OF) +/* + * This struct is just used as place holder. It will be filled with + * data from dt node + */ +static struct snd_soc_dai_link evm_dai = { +}; + +/* davinci evm audio machine driver */ +static struct snd_soc_card evm_soc_card = { + .owner = THIS_MODULE, + .dai_link = &evm_dai, + .num_links = 1, +}; + +static int davinci_evm_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + const char *stringPtr, *propname; + u32 val; + int ret = 0; + + propname = "dai-data"; + val = of_property_count_strings(np, propname); + if (val < 0) + return val; + + ret = of_property_read_string_index(np, propname, 0, &evm_dai.name); + if (ret < 0) + return ret; + + ret = of_property_read_string_index(np, propname, 1, + &evm_dai.stream_name); + if (ret < 0) + return ret; + + ret = of_property_read_string_index(np, propname, 2, + &evm_dai.codec_dai_name); + if (ret < 0) + return ret; + + ret = of_property_read_string_index(np, propname, 3, &stringPtr); + if (ret < 0) + return ret; + else + if (strcasecmp(stringPtr, "true") == 0) + evm_dai.init = evm_aic3x_init; + + ret = of_property_read_string_index(np, propname, 4, &stringPtr); + if (ret < 0) { + return ret; + } else { + if (strcasecmp(stringPtr, "evm-ops") == 0) + evm_dai.ops = &evm_ops; + else if (strcasecmp(stringPtr, "evm-spdif-ops") == 0) + evm_dai.ops = &evm_spdif_ops; + } + + evm_dai.codec_of_node = of_parse_phandle(np, "ti,audio-codec", 0); + if (!evm_dai.codec_of_node) + return -EINVAL; + + evm_dai.cpu_dai_of_node = of_parse_phandle(np, + "ti,mcasp-controller", 0); + if (!evm_dai.cpu_dai_of_node) + return -EINVAL; + + evm_dai.platform_of_node = evm_dai.cpu_dai_of_node; + + evm_soc_card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&evm_soc_card, "ti,model"); + if (ret) + return ret; + + ret = snd_soc_register_card(&evm_soc_card); + if (ret) + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + + return ret; +} + +static int __devexit davinci_evm_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + + return 0; +} + +static const struct of_device_id davinci_evm_dt_ids[] = { + { .compatible = "ti,davinci-evm-audio", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, davinci_mcasp_dt_ids); + +static struct platform_driver davinci_evm_driver = { + .probe = davinci_evm_probe, + .remove = __devexit_p(davinci_evm_remove), + .driver = { + .name = "davinci_evm", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(davinci_evm_dt_ids), + }, +}; +#endif + static struct platform_device *evm_snd_device; static int __init evm_init(void) @@ -289,6 +416,14 @@ static int __init evm_init(void) struct snd_soc_card *evm_snd_dev_data; int index; int ret; +#if defined(CONFIG_OF) + struct device_node *np; + + np = of_find_compatible_node(NULL, NULL, "ti,davinci-evm-audio"); + if (np) { + return platform_driver_register(&davinci_evm_driver); + } +#endif if (machine_is_davinci_evm()) { evm_snd_dev_data = &dm6446_snd_soc_card_evm; @@ -325,6 +460,16 @@ static int __init evm_init(void) static void __exit evm_exit(void) { +#if defined(CONFIG_OF) + struct device_node *np; + + np = of_find_compatible_node(NULL, NULL, "ti,davinci-evm-audio"); + if (np) { + platform_driver_unregister(&davinci_evm_driver); + return; + } +#endif + platform_device_unregister(evm_snd_device); } -- 1.7.1

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[alsa-devel] [PATCH 05/22] ASoC: Ux500: Enable MOP500 driver for Device Tree
by Lee Jones 10 Sep '12

10 Sep '12

Here we ensure that the MOP500 audio driver will be probed during a Device Tree boot. We also parse the sound node to link together the codec, dma and the CPU-side Digital Audio Interface. CC: alsa-devel(a)alsa-project.org Signed-off-by: Lee Jones <lee.jones(a)linaro.org> --- sound/soc/ux500/mop500.c | 40 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 40 insertions(+) diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c index 31c4d26..6840df7 100644 --- a/sound/soc/ux500/mop500.c +++ b/sound/soc/ux500/mop500.c @@ -16,6 +16,7 @@ #include <linux/module.h> #include <linux/io.h> #include <linux/spi/spi.h> +#include <linux/of.h> #include <sound/soc.h> #include <sound/initval.h> @@ -56,8 +57,35 @@ static struct snd_soc_card mop500_card = { .num_links = ARRAY_SIZE(mop500_dai_links), }; +static int __devinit mop500_of_probe(struct platform_device *pdev, + struct device_node *np) +{ + struct device_node *codec_np, *msp_np[2]; + int i; + + msp_np[0] = of_parse_phandle(np, "stericsson,cpu-dai", 0); + msp_np[1] = of_parse_phandle(np, "stericsson,cpu-dai", 1); + codec_np = of_parse_phandle(np, "stericsson,audio-codec", 0); + + if (!(msp_np[0] && msp_np[1] && codec_np)) { + dev_err(&pdev->dev, "Phandle missing or invalid\n"); + return -EINVAL; + } + + for (i = 0; i < 2; i++) { + mop500_dai_links[i].cpu_of_node = msp_np[i]; + mop500_dai_links[i].cpu_dai_name = NULL; + mop500_dai_links[i].codec_of_node = codec_np; + mop500_dai_links[i].codec_name = NULL; + } + + snd_soc_of_parse_card_name(&mop500_card, "stericsson,card-name"); + + return 0; +} static int __devinit mop500_probe(struct platform_device *pdev) { + struct device_node *np = pdev->dev.of_node; int ret; pr_debug("%s: Enter.\n", __func__); @@ -66,6 +94,12 @@ static int __devinit mop500_probe(struct platform_device *pdev) mop500_card.dev = &pdev->dev; + if (np) { + ret = mop500_of_probe(pdev, np); + if (ret) + return ret; + } + dev_dbg(&pdev->dev, "%s: Card %s: Set platform drvdata.\n", __func__, mop500_card.name); platform_set_drvdata(pdev, &mop500_card); @@ -101,10 +135,16 @@ static int __devexit mop500_remove(struct platform_device *pdev) return 0; } +static const struct of_device_id snd_soc_mop500_match[] = { + { .compatible = "stericsson,snd-soc-mop500", }, + {}, +}; + static struct platform_driver snd_soc_mop500_driver = { .driver = { .owner = THIS_MODULE, .name = "snd-soc-mop500", + .of_match_table = snd_soc_mop500_match, }, .probe = mop500_probe, .remove = __devexit_p(mop500_remove), -- 1.7.9.5

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[alsa-devel] [PATCH 06/22] ASoC: Ux500: Enable ux500 MSP driver for Device Tree
by Lee Jones 10 Sep '12

10 Sep '12

Register both parts of the MSP driver from Device Tree so that they are probed when Device Tree is enabled. Also, as there is platform data involved, we ensure that there is allocated memory to place the configuration into and that the correct information is extracted from the DT binary. CC: alsa-devel(a)alsa-project.org Signed-off-by: Lee Jones <lee.jones(a)linaro.org> --- sound/soc/ux500/ux500_msp_dai.c | 6 ++++++ sound/soc/ux500/ux500_msp_i2s.c | 22 +++++++++++++++++++--- 2 files changed, 25 insertions(+), 3 deletions(-) diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index 772cb19..0f7dd49 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -833,10 +833,16 @@ static int __devexit ux500_msp_drv_remove(struct platform_device *pdev) return 0; } +static const struct of_device_id ux500_msp_i2c_match[] = { + { .compatible = "stericsson,ux500-msp-i2s", }, + {}, +}; + static struct platform_driver msp_i2s_driver = { .driver = { .name = "ux500-msp-i2s", .owner = THIS_MODULE, + .of_match_table = ux500_msp_i2c_match, }, .probe = ux500_msp_drv_probe, .remove = ux500_msp_drv_remove, diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c index 2cbfc54..5e0bf8c 100644 --- a/sound/soc/ux500/ux500_msp_i2s.c +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -18,6 +18,7 @@ #include <linux/pinctrl/consumer.h> #include <linux/delay.h> #include <linux/slab.h> +#include <linux/of.h> #include <mach/hardware.h> #include <mach/msp.h> @@ -692,16 +693,31 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev, int ret = 0; struct resource *res = NULL; struct i2s_controller *i2s_cont; + struct device_node *np = pdev->dev.of_node; struct ux500_msp *msp; - dev_dbg(&pdev->dev, "%s: Enter (name: %s, id: %d).\n", __func__, - pdev->name, platform_data->id); - *msp_p = devm_kzalloc(&pdev->dev, sizeof(struct ux500_msp), GFP_KERNEL); msp = *msp_p; if (!msp) return -ENOMEM; + if (np) { + if (!platform_data) { + platform_data = devm_kzalloc(&pdev->dev, + sizeof(struct msp_i2s_platform_data), GFP_KERNEL); + if (!platform_data) + ret = -ENOMEM; + } + } else + if (!platform_data) + ret = -EINVAL; + + if (ret) + goto err_res; + + dev_dbg(&pdev->dev, "%s: Enter (name: %s, id: %d).\n", __func__, + pdev->name, platform_data->id); + msp->id = platform_data->id; msp->dev = &pdev->dev; msp->dma_cfg_rx = platform_data->msp_i2s_dma_rx; -- 1.7.9.5

2 2

[alsa-devel] [RFC PATCH 1/11] expose sound device topology information
by Clemens Ladisch 07 Sep '12

07 Sep '12

Hi, these are some tentative patches to allow sound drivers to provide routing information to userspace. (So far, snd-usb-audio has a partial implementation; I haven't yet had time for snd-hda-intel. I haven't even looked at ASoC.) The media controller API <linux/media.h> allows prividing routing information, but its implementation does not quite fit the sound drivers: * it allows reconfiguration of links, but not in a way that would be useful for ALSA selector controls; * it allows drivers to access the entity graph and pipelines, which is not needed; * it is rather heavyweight. Therefore, these patches create their own implementation of the media controller API. In this implementation, all entity and link information is completely static, and more space-efficient. For simplicity, the ioctls hook into the ALSA control device instead of creating a new /dev/media* device. The separate implementation does not allow sharing of one media controller device in the case of combined audio/video devices. However, separate drivers would already be a problem for HDMI outputs (GPU and sound); it appears we might need some mechanism to connect the topologies of multiple media devices. TLVs (for jack entities, and for associating controls with entities) are not yet defined. Example output for my simple UA-1A device (not all information is decoded correctly): $ media-ctl -p -d /dev/snd/controlC5 Opening media device /dev/snd/controlC5 Enumerating entities Found 4 entities Enumerating pads and links Media controller API version 0.0.0 Media device information ------------------------ driver snd-usb-audio model EDIROL UA-1A serial bus info usb-0000:00:12.2-4.3 hw revision 0x101 driver version 0.0.0 Device topology - entity 1: USB Audio (1 pad, 1 link) type Node subtype ALSA device node name /dev/tty (should be hw:5,0,0) pad0: Source -> "OT3":0 [ENABLED,IMMUTABLE] - entity 3: OT3 (1 pad, 1 link) type Unknown subtype Unknown (should be Jack) pad0: Sink <- "USB Audio":0 [ENABLED,IMMUTABLE] - entity 4: IT4 (1 pad, 1 link) type Unknown subtype Unknown pad0: Source -> "USB Audio":0 [ENABLED,IMMUTABLE] - entity 7: USB Audio (1 pad, 1 link) type Node subtype ALSA device node name /dev/tty pad0: Sink <- "IT4":0 [ENABLED,IMMUTABLE] include/linux/media.h | 9 include/sound/core.h | 13 + include/sound/media.h | 116 ++++++++++ include/sound/pcm.h | 6 sound/core/Kconfig | 7 sound/core/Makefile | 1 sound/core/init.c | 4 sound/core/media.c | 425 ++++++++++++++++++++++++++++++++++++++ sound/core/pcm.c | 46 ++++ sound/core/sound.c | 3 sound/pci/hda/hda_intel.c | 2 sound/usb/card.c | 2 sound/usb/mixer.c | 278 ++++++++++++++++++++++++ sound/usb/quirks.c | 4 sound/usb/stream.c | 16 - sound/usb/stream.h | 1 16 files changed, 925 insertions(+), 8 deletions(-) Regards, Clemens

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[alsa-devel] [PATCH] ASoC: SAMSUNG: Add SND_SOC_DAIFMT_CONT option for snd_soc_set_fmt()
by 박상수 06 Sep '12

06 Sep '12

SND_SOC_DAIFMT_CONT option is now needed because of below commit: ASoC: Allow DAI formats to be specified in the dai_link Otherwise, s3c_pcm_set_fmt() returns -EINVAL. Signed-off-by: Sangsu Park <sangsu4u.park(a)samsung.com> --- sound/soc/samsung/smdk_wm8994pcm.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/samsung/smdk_wm8994pcm.c b/sound/soc/samsung/smdk_wm8994pcm.c index 77ecba9..cb88cbc 100644 --- a/sound/soc/samsung/smdk_wm8994pcm.c +++ b/sound/soc/samsung/smdk_wm8994pcm.c @@ -77,6 +77,7 @@ static int smdk_wm8994_pcm_hw_params(struct snd_pcm_substream *substream, /* Set the cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_B + | SND_SOC_DAIFMT_CONT | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) -- 1.7.1

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[alsa-devel] radeon multi-channel lpcm
by Pierre Ossman 06 Sep '12

06 Sep '12

Hi, I figured I'd have a go at finishing up support for multi-channel lcpm over HDMI on radeon cards. I read a post from the AMD guys stating that the GPU portions were already in place, and it was the audio pieces that were missing. Unfortunately I was unable to figure out exactly what it is that remains to be done. I poked around in patch_hdmi.c, and concluded that the generic code is what's being used for modern radeon cards. The card exposes two widgets (and output widget and a pin complex). But both widgets indicate through their capability registers that they only support stereo. I don't know enough about HDA to have any clue where to go from here. Should the capability field be ignored for HDMI? Or does it need some coaxing to allow more than stereo? The ELD stuff isn't properly updated from the radeon driver, but the code seems to suggest it is just for hints, and not critical. Rgds -- -- Pierre Ossman WARNING: This correspondence is being monitored by FRA, a Swedish intelligence agency. Make sure your server uses encryption for SMTP traffic and consider using PGP for end-to-end encryption.

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[alsa-devel] [PATCH] ASoC: omap-mcbsp: Fix compilation error due to leftover code
by Peter Ujfalusi 05 Sep '12

05 Sep '12

Part of commit (which patches sound/soc/omap/mcbsp.c file): 8fef626 ARM/ASoC: omap-mcbsp: Remove CLKR/FSR mux configuration code since the tree where it has been applied did not had the earlier patch: d0db84e ASoC: omap-mcbsp: Fix 6pin mux configuration which changed code around omap_mcbsp_6pin_src_mux(). Because of the missing part from 8fef626 the sound/soc/omap/mcbsp.c does not compile in linux-next. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi Mark, This patch has been generated against linux-next as of today. The reason for this is that that is the only place where we have the upstream patch (in 3.6-rc3): d0db84e ASoC: omap-mcbsp: Fix 6pin mux configuration and the 8fef626 ARM/ASoC: omap-mcbsp: Remove CLKR/FSR mux configuration code commit in the same branch. When you applied 8fef626 you mentioned that it did not applied cleanly. The reason was that the branch did not had commit d0db84e applied prior to the series: http://mailman.alsa-project.org/pipermail/alsa-devel/2012-August/054668.html Now I can not find the d0db84e commit in the asoc tree (not in for-3.6, for-3.7 and not in for-next branch) anymore. I don't really know what has happened, but as of now this patch is needed to fix the OMAP audio in linux-next. I have also prepared a branch with the OMAP patches [1]: topic/omap rebased on top of today's mainline and fixed up the 8fef626 to it's original content. But I think it is already late to revert the OMAP McBSP patches from for-next and from for-3.7 branches... [1] git://gitorious.org/omap-audio/linux-audio.git peter/topic/for-mark/omap-3.7 CC-ing linux-next as well. Regards, Peter sound/soc/omap/mcbsp.c | 31 ------------------------------- 1 file changed, 31 deletions(-) diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 935ccf6..bc06175 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -762,37 +762,6 @@ int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id) } -int omap_mcbsp_6pin_src_mux(struct omap_mcbsp *mcbsp, u8 mux) -{ - const char *signal, *src; - - if (!mcbsp->pdata->mux_signal) - return -EINVAL; - - switch (mux) { - case CLKR_SRC_CLKR: - signal = "clkr"; - src = "clkr"; - break; - case CLKR_SRC_CLKX: - signal = "clkr"; - src = "clkx"; - break; - case FSR_SRC_FSR: - signal = "fsr"; - src = "fsr"; - break; - case FSR_SRC_FSX: - signal = "fsr"; - src = "fsx"; - break; - default: - return -EINVAL; - } - - return mcbsp->pdata->mux_signal(mcbsp->dev, signal, src); -} - #define max_thres(m) (mcbsp->pdata->buffer_size) #define valid_threshold(m, val) ((val) <= max_thres(m)) #define THRESHOLD_PROP_BUILDER(prop) \ -- 1.7.12

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[alsa-devel] [asoc:for-3.7 95/95] sound/soc/codecs/wm0010.c:850:7-27: ERROR: Threaded IRQ with no primary handler requested without IRQF_ONESHOT
by Fengguang Wu 05 Sep '12

05 Sep '12

[add more CC on request of Dimitris] Hi Dimitris, Mark, FYI, there are new coccinelle warnings show up in tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-3.7 head: e3523e01869da20fdd12ffd19ae1df7bf492650e commit: e3523e01869da20fdd12ffd19ae1df7bf492650e [95/95] ASoC: wm0010: Add initial wm0010 DSP driver All coccinelle warnings: + sound/soc/codecs/wm0010.c:850:7-27: ERROR: Threaded IRQ with no primary handler requested without IRQF_ONESHOT -- + sound/soc/codecs/wm0010.c:660:1-7: preceding lock on line 359 vim +850 sound/soc/codecs/wm0010.c 847 trigger = IRQF_TRIGGER_FALLING; 848 trigger |= IRQF_ONESHOT; 849 > 850 ret = request_threaded_irq(irq, NULL, wm0010_irq, trigger, 851 "wm0010", wm0010); 852 if (ret) 853 dev_err(wm0010->dev, "Failed to request IRQ %d: %d\n", Please consider folding the attached diff :-) --- 0-DAY kernel build testing backend Open Source Technology Centre Fengguang Wu <wfg(a)linux.intel.com> Intel Corporation

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[alsa-devel] [PATCH] ALSA: HDMI - Enable HBR feature on Intel chips
by Wang Xingchao 05 Sep '12

05 Sep '12

HDMI channel remapping apparently effects HBR packets on Intel's chips. For compressed non-PCM audio, use "straight-through" channel mapping. For uncompressed multi-channel pcm audio, use normal mapping. Signed-off-by: Wang Xingchao <xingchao.wang(a)intel.com> Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> --- sound/pci/hda/patch_hdmi.c | 50 +++++++++++++++++++++++++++----------------- 1 file changed, 31 insertions(+), 19 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index d9439c5..fda89e3 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -231,6 +231,11 @@ static struct cea_channel_speaker_allocation channel_allocations[] = { { .ca_index = 0x01, .speakers = { 0, 0, 0, 0, 0, LFE, FR, FL } }, /* Dolby Surround */ { .ca_index = 0x02, .speakers = { 0, 0, 0, 0, FC, 0, FR, FL } }, +{ .ca_index = 0x03, .speakers = { 0, 0, 0, 0, FC, LFE, FR, FL } }, +{ .ca_index = 0x04, .speakers = { 0, 0, 0, RC, 0, 0, FR, FL } }, +{ .ca_index = 0x05, .speakers = { 0, 0, 0, RC, 0, LFE, FR, FL } }, +{ .ca_index = 0x06, .speakers = { 0, 0, 0, RC, FC, 0, FR, FL } }, +{ .ca_index = 0x07, .speakers = { 0, 0, 0, RC, FC, LFE, FR, FL } }, /* surround40 */ { .ca_index = 0x08, .speakers = { 0, 0, RR, RL, 0, 0, FR, FL } }, /* surround41 */ @@ -239,22 +244,15 @@ static struct cea_channel_speaker_allocation channel_allocations[] = { { .ca_index = 0x0a, .speakers = { 0, 0, RR, RL, FC, 0, FR, FL } }, /* surround51 */ { .ca_index = 0x0b, .speakers = { 0, 0, RR, RL, FC, LFE, FR, FL } }, - /* 6.1 */ -{ .ca_index = 0x0f, .speakers = { 0, RC, RR, RL, FC, LFE, FR, FL } }, - /* surround71 */ -{ .ca_index = 0x13, .speakers = { RRC, RLC, RR, RL, FC, LFE, FR, FL } }, - -{ .ca_index = 0x03, .speakers = { 0, 0, 0, 0, FC, LFE, FR, FL } }, -{ .ca_index = 0x04, .speakers = { 0, 0, 0, RC, 0, 0, FR, FL } }, -{ .ca_index = 0x05, .speakers = { 0, 0, 0, RC, 0, LFE, FR, FL } }, -{ .ca_index = 0x06, .speakers = { 0, 0, 0, RC, FC, 0, FR, FL } }, -{ .ca_index = 0x07, .speakers = { 0, 0, 0, RC, FC, LFE, FR, FL } }, { .ca_index = 0x0c, .speakers = { 0, RC, RR, RL, 0, 0, FR, FL } }, { .ca_index = 0x0d, .speakers = { 0, RC, RR, RL, 0, LFE, FR, FL } }, { .ca_index = 0x0e, .speakers = { 0, RC, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x0f, .speakers = { 0, RC, RR, RL, FC, LFE, FR, FL } }, { .ca_index = 0x10, .speakers = { RRC, RLC, RR, RL, 0, 0, FR, FL } }, { .ca_index = 0x11, .speakers = { RRC, RLC, RR, RL, 0, LFE, FR, FL } }, { .ca_index = 0x12, .speakers = { RRC, RLC, RR, RL, FC, 0, FR, FL } }, + /* surround71 */ +{ .ca_index = 0x13, .speakers = { RRC, RLC, RR, RL, FC, LFE, FR, FL } }, { .ca_index = 0x14, .speakers = { FRC, FLC, 0, 0, 0, 0, FR, FL } }, { .ca_index = 0x15, .speakers = { FRC, FLC, 0, 0, 0, LFE, FR, FL } }, { .ca_index = 0x16, .speakers = { FRC, FLC, 0, 0, FC, 0, FR, FL } }, @@ -537,22 +535,36 @@ static void hdmi_debug_channel_mapping(struct hda_codec *codec, static void hdmi_setup_channel_mapping(struct hda_codec *codec, hda_nid_t pin_nid, + hda_nid_t cvt_nid, int ca) { int i; int err; - - if (hdmi_channel_mapping[ca][1] == 0) { - for (i = 0; i < channel_allocations[ca].channels; i++) - hdmi_channel_mapping[ca][i] = i | (i << 4); + unsigned short val; + int tmp_mapping[8]; + bool non_pcm = false; + + val = snd_hda_codec_read(codec, cvt_nid, 0, AC_VERB_GET_DIGI_CONVERT_1, 0); + non_pcm = !!(val & AC_DIG1_NONAUDIO); + + if (hdmi_channel_mapping[ca][1] == 0 || non_pcm) { + for (i = 0; i < channel_allocations[ca].channels; i++) { + if (non_pcm) + tmp_mapping[i] = i | (i << 4); + else + hdmi_channel_mapping[ca][i] = i | (i << 4); + } for (; i < 8; i++) - hdmi_channel_mapping[ca][i] = 0xf | (i << 4); + if (non_pcm) + tmp_mapping[i] = 0xf | (i << 4); + else + hdmi_channel_mapping[ca][i] = 0xf | (i << 4); } for (i = 0; i < 8; i++) { err = snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_CHAN_SLOT, - hdmi_channel_mapping[ca][i]); + non_pcm ? tmp_mapping[i] : hdmi_channel_mapping[ca][i]); if (err) { snd_printdd(KERN_NOTICE "HDMI: channel mapping failed\n"); @@ -686,7 +698,7 @@ static bool hdmi_infoframe_uptodate(struct hda_codec *codec, hda_nid_t pin_nid, } static void hdmi_setup_audio_infoframe(struct hda_codec *codec, int pin_idx, - struct snd_pcm_substream *substream) + hda_nid_t cvt_nid, struct snd_pcm_substream *substream) { struct hdmi_spec *spec = codec->spec; struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx]; @@ -731,13 +743,13 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, int pin_idx, * sizeof(*dp_ai) to avoid partial match/update problems when * the user switches between HDMI/DP monitors. */ + hdmi_setup_channel_mapping(codec, pin_nid, cvt_nid, ca); if (!hdmi_infoframe_uptodate(codec, pin_nid, ai.bytes, sizeof(ai))) { snd_printdd("hdmi_setup_audio_infoframe: " "pin=%d channels=%d\n", pin_nid, channels); - hdmi_setup_channel_mapping(codec, pin_nid, ca); hdmi_stop_infoframe_trans(codec, pin_nid); hdmi_fill_audio_infoframe(codec, pin_nid, ai.bytes, sizeof(ai)); @@ -1151,7 +1163,7 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, hdmi_set_channel_count(codec, cvt_nid, substream->runtime->channels); - hdmi_setup_audio_infoframe(codec, pin_idx, substream); + hdmi_setup_audio_infoframe(codec, pin_idx, cvt_nid, substream); pinctl = snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); -- 1.7.9.5

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Re: [alsa-devel] How to define card specific pcm devices?
by Raymond Yau 05 Sep '12

05 Sep '12

2012-8-17 下午10:51 於 "Tanu Kaskinen" <tanu.kaskinen(a)digia.com> 寫道： > > Ping? > > If card specific pcm definitions are not possible, would it be > acceptable to have a generic "front_left:CARD" or "first_channel:CARD" > pcm, which would transform "front:CARD" or "hw:CARD" into a mono device? As pulseaudio server use SND_PCM_NO_AUTO_CHANNEL which diable channel conversion, this left channel device only work with other alsa applications except pulseaudio when your sound card does not support mono. http://cgit.freedesktop.org/pulseaudio/pulseaudio/commit/?id=ff4af902cf4ac0… the solution is to change the mixing in pulseaudio by using left channel for this mic and those stereo mic with inverted phase , and average for mono playback device . stereo mic and line-in how about the 4 channel mic? > > -- Tanu > > > On Fri, 2012-08-10 at 14:25 +0300, Tanu Kaskinen wrote: > > Hi, > > > > I'm writing UCM configuration for RX-51 (Nokia N900), and I have a > > problem: "hw:RX51" can't be opened in mono mode, but there's only one > > integrated mono microphone, and its audio is routed to the left input > > channel. Applications will get only silence in the right channel when > > they use "hw:RX51". > > > > The best solution that I have come up is to define a pcm that converts > > "hw:RX51" into a mono device. The problem is that I don't know how > > exactly to do that. I guess the route plugin can be used for that: > > > > pcm.rx51_mono_from_left { > > type route > > slave.pcm hw:RX51 > > slave.channels 2 > > ttable.0.0 1 > > } > > > > I tried putting that definition in /usr/share/alsa/cards/RX-51.conf (and > > also RX51.conf, since it's not entirely clear to me when the dash should > > be omitted). I have this in the UCM configuration: > > > > Value { > > CapturePCM "rx51_mono_from_left" > > CaptureChannels "1" > > } > > > > This is printed to the PulseAudio log: > > > > (alsa-lib)pcm.c: Unknown PCM rx51_mono_from_left > > > > So the rx51_mono_from_left definition is not loaded. I'm not terribly > > surprised: I guess the card configuration files are loaded only when > > necessary, and when PulseAudio tries to open "rx51_mono_from_left", > > alsa-lib can't really know that it's somehow related to the RX-51 card. > > > > How should I solve this? > > > >

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