January 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH - [hdspmixer] 1/1] hdspmixer: Don't use channelmap from hdspm.h
by Adrian Knoth 27 Jan '11

27 Jan '11

The channel mapping has been moved to hdspm.c, so it's no longer available to userspace tools. For now, let's simply copy (duplicate) the data and wait for a way to query this information from the driver. Signed-off-by: Adrian Knoth <adi(a)drcomp.erfurt.thur.de> diff --git a/hdspmixer/src/HDSPMixerCard.h b/hdspmixer/src/HDSPMixerCard.h index 9303d21..cfb2402 100644 --- a/hdspmixer/src/HDSPMixerCard.h +++ b/hdspmixer/src/HDSPMixerCard.h @@ -28,6 +28,7 @@ #include <alsa/sound/hdsp.h> #include <alsa/sound/hdspm.h> #include "defines.h" +#include "channelmap.h" #include "HDSPMixerWindow.h" class HDSPMixerWindow; diff --git a/hdspmixer/src/channelmap.h b/hdspmixer/src/channelmap.h new file mode 100644 index 0000000..de89140 --- /dev/null +++ b/hdspmixer/src/channelmap.h @@ -0,0 +1,167 @@ +#ifndef channelmap_H +#define channelmap_H + +static char channel_map_unity_ss[HDSPM_MAX_CHANNELS] = { + 0, 1, 2, 3, 4, 5, 6, 7, + 8, 9, 10, 11, 12, 13, 14, 15, + 16, 17, 18, 19, 20, 21, 22, 23, + 24, 25, 26, 27, 28, 29, 30, 31, + 32, 33, 34, 35, 36, 37, 38, 39, + 40, 41, 42, 43, 44, 45, 46, 47, + 48, 49, 50, 51, 52, 53, 54, 55, + 56, 57, 58, 59, 60, 61, 62, 63 +}; + +static char channel_map_unity_ds[HDSPM_MAX_CHANNELS] = { + 0, 2, 4, 6, 8, 10, 12, 14, + 16, 18, 20, 22, 24, 26, 28, 30, + 32, 34, 36, 38, 40, 42, 44, 46, + 48, 50, 52, 54, 56, 58, 60, 62, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, +}; + +static char channel_map_unity_qs[HDSPM_MAX_CHANNELS] = { + 0, 4, 8, 12, 16, 20, 24, 28, + 32, 36, 40, 44, 48, 52, 56, 60, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, +}; + +static char channel_map_raydat_ss[HDSPM_MAX_CHANNELS] = { + 4, 5, 6, 7, 8, 9, 10, 11, /* ADAT 1 */ + 12, 13, 14, 15, 16, 17, 18, 19, /* ADAT 2 */ + 20, 21, 22, 23, 24, 25, 26, 27, /* ADAT 3 */ + 28, 29, 30, 31, 32, 33, 34, 35, /* ADAT 4 */ + 0, 1, /* AES */ + 2, 3, /* SPDIF */ + -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, +}; + +static char channel_map_raydat_ds[HDSPM_MAX_CHANNELS] = { + 4, 5, 6, 7, /* ADAT 1 */ + 8, 9, 10, 11, /* ADAT 2 */ + 12, 13, 14, 15, /* ADAT 3 */ + 16, 17, 18, 19, /* ADAT 4 */ + 0, 1, /* AES */ + 2, 3, /* SPDIF */ + -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, +}; + +static char channel_map_raydat_qs[HDSPM_MAX_CHANNELS] = { + 4, 5, /* ADAT 1 */ + 6, 7, /* ADAT 2 */ + 8, 9, /* ADAT 3 */ + 10, 11, /* ADAT 4 */ + 0, 1, /* AES */ + 2, 3, /* SPDIF */ + -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, +}; + +static char channel_map_aio_in_ss[HDSPM_MAX_CHANNELS] = { + 0, 1, /* line in */ + 8, 9, /* aes in, */ + 10, 11, /* spdif in */ + 12, 13, 14, 15, 16, 17, 18, 19, /* ADAT in */ + -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, +}; + +static char channel_map_aio_out_ss[HDSPM_MAX_CHANNELS] = { + 0, 1, /* line out */ + 8, 9, /* aes out */ + 10, 11, /* spdif out */ + 12, 13, 14, 15, 16, 17, 18, 19, /* ADAT out */ + 6, 7, /* phone out */ + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, +}; + +static char channel_map_aio_in_ds[HDSPM_MAX_CHANNELS] = { + 0, 1, /* line in */ + 8, 9, /* aes in */ + 10, 11, /* spdif in */ + 12, 14, 16, 18, /* adat in */ + -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1 +}; + +static char channel_map_aio_out_ds[HDSPM_MAX_CHANNELS] = { + 0, 1, /* line out */ + 8, 9, /* aes out */ + 10, 11, /* spdif out */ + 12, 14, 16, 18, /* adat out */ + 6, 7, /* phone out */ + -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1 +}; + +static char channel_map_aio_in_qs[HDSPM_MAX_CHANNELS] = { + 0, 1, /* line in */ + 8, 9, /* aes in */ + 10, 11, /* spdif in */ + 12, 16, /* adat in */ + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1 +}; + +static char channel_map_aio_out_qs[HDSPM_MAX_CHANNELS] = { + 0, 1, /* line out */ + 8, 9, /* aes out */ + 10, 11, /* spdif out */ + 12, 16, /* adat out */ + 6, 7, /* phone out */ + -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1 +}; + +#endif /* channelmap_H */ -- 1.7.2.3

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[alsa-devel] [PATCH 1/4] ASoC: soc-core: Increment codec and platform driver refcounts before probing
by Jarkko Nikula 27 Jan '11

27 Jan '11

Commit f6c2ed5 "ASoC: Fix the device references to codec and platform drivers" moved codec and platform driver refcount increments from soc_bind_dai_link to more appropriate places. Adjust a little them so that refcounts are incremented before executing the driver probe functions. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> Cc: Vinod Koul <vinod.koul(a)intel.com> Cc: Harsha Priya <priya.harsha(a)intel.com> --- I don't know any regression with this so probably 2.6.39 material. --- sound/soc/soc-core.c | 21 +++++++++++++-------- 1 files changed, 13 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 14861f9..4da185f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1408,26 +1408,31 @@ static int soc_probe_codec(struct snd_soc_card *card, codec->dapm.card = card; soc_set_name_prefix(card, codec); + if (!try_module_get(codec->dev->driver->owner)) + return -ENODEV; + if (codec->driver->probe) { ret = codec->driver->probe(codec); if (ret < 0) { dev_err(codec->dev, "asoc: failed to probe CODEC %s: %d\n", codec->name, ret); - return ret; + goto err_probe; } } soc_init_codec_debugfs(codec); /* mark codec as probed and add to card codec list */ - if (!try_module_get(codec->dev->driver->owner)) - return -ENODEV; - codec->probed = 1; list_add(&codec->card_list, &card->codec_dev_list); list_add(&codec->dapm.list, &card->dapm_list); + return 0; + +err_probe: + module_put(codec->dev->driver->owner); + return ret; } @@ -1545,19 +1550,19 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) /* probe the platform */ if (!platform->probed) { + if (!try_module_get(platform->dev->driver->owner)) + return -ENODEV; + if (platform->driver->probe) { ret = platform->driver->probe(platform); if (ret < 0) { printk(KERN_ERR "asoc: failed to probe platform %s\n", platform->name); + module_put(platform->dev->driver->owner); return ret; } } /* mark platform as probed and add to card platform list */ - - if (!try_module_get(platform->dev->driver->owner)) - return -ENODEV; - platform->probed = 1; list_add(&platform->card_list, &card->platform_dev_list); } -- 1.7.2.3

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[alsa-devel] Triggering a DAPM widget event
by Harrod, John 27 Jan '11

27 Jan '11

Hi all, I have a program that uses the ALSA PCM interface to play sounds on an embedded system. Does anyone know how alsa-lib triggers a SND_SOC_DAPM_PRE_PMD widget event in DAPM core? On my system the event only gets called when the pcm is closed with snd_pcm_close(). Is there another function that can cause a widget shutdown? Or is opening and closing the PCM every time I want to play a sound a good practice for saving power? Regards, John ________________________________ The information contained in this message may be confidential and legally protected under applicable law. The message is intended solely for the addressee(s). If you are not the intended recipient, you are hereby notified that any use, forwarding, dissemination, or reproduction of this message is strictly prohibited and may be unlawful. If you are not the intended recipient, please contact the sender by return e-mail and destroy all copies of the original message.

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[PATCH] ALSA: HDA: Fix microphone(s) on Lenovo Edge 13
by David Henningsson 27 Jan '11

27 Jan '11

BugLink: http://bugs.launchpad.net/bugs/708521 This Edge 13 model has an internal mic at 0x1a and should therefore use the asus quirk. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 9867afc..7e1ca43 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3120,6 +3120,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */ -- 1.7.1 --------------020306060702000000060003--

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[alsa-devel] [PATCH 2/2] ALSA: HDA: Add a new model "asus" for Conexant 5066/205xx
by David Henningsson 27 Jan '11

27 Jan '11

BugLink: http://bugs.launchpad.net/bugs/701271 This new model, named "asus", is identical to the "hp_laptop" model, except for the location of the internal mic, which is at pin 0x1a. It is used for Asus K52JU and Lenovo G560. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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[alsa-devel] [PATCH 2.6.38] ASoC: Fix module refcount for auxiliary devices
by Jarkko Nikula 26 Jan '11

26 Jan '11

Commit f6c2ed5 "ASoC: Fix the device references to codec and platform drivers" moved codec driver refcount increments from soc_bind_dai_link into soc_probe_codec. However, the commit didn't remove try_module_get from soc_probe_aux_dev so the auxiliary device reference counts are incremented twice as the soc_probe_codec is called from soc_probe_aux_dev too. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> Cc: Vinod Koul <vinod.koul(a)intel.com> Cc: Harsha Priya <priya.harsha(a)intel.com> --- sound/soc/soc-core.c | 3 --- 1 files changed, 0 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index bac7291..c4b6061 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1664,9 +1664,6 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num) goto out; found: - if (!try_module_get(codec->dev->driver->owner)) - return -ENODEV; - ret = soc_probe_codec(card, codec); if (ret < 0) return ret; -- 1.7.2.3

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[alsa-devel] [PATCH 1/3] ASoC: Use card rather than soc-audio device to card PM functions
by Mark Brown 26 Jan '11

26 Jan '11

The platform device for the card is tied closely to the soc-audio implementation which we're currently trying to remove in favour of allowing cards to have their own devices. Begin removing it by replacing it with the card in the suspend and resume callbacks we give to cards, also taking the opportunity to remove the legacy suspend types which are currently hard coded anyway. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 8 ++++---- sound/soc/pxa/raumfeld.c | 4 ++-- sound/soc/pxa/zylonite.c | 5 ++--- sound/soc/soc-core.c | 9 ++++----- 4 files changed, 12 insertions(+), 14 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 00c6fba..d1cc1a2 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -655,10 +655,10 @@ struct snd_soc_card { /* the pre and post PM functions are used to do any PM work before and * after the codec and DAI's do any PM work. */ - int (*suspend_pre)(struct platform_device *pdev, pm_message_t state); - int (*suspend_post)(struct platform_device *pdev, pm_message_t state); - int (*resume_pre)(struct platform_device *pdev); - int (*resume_post)(struct platform_device *pdev); + int (*suspend_pre)(struct snd_soc_card *card); + int (*suspend_post)(struct snd_soc_card *card); + int (*resume_pre)(struct snd_soc_card *card); + int (*resume_post)(struct snd_soc_card *card); /* callbacks */ int (*set_bias_level)(struct snd_soc_card *, diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c index 0fd60f4..db1dd56 100644 --- a/sound/soc/pxa/raumfeld.c +++ b/sound/soc/pxa/raumfeld.c @@ -151,13 +151,13 @@ static struct snd_soc_ops raumfeld_cs4270_ops = { .hw_params = raumfeld_cs4270_hw_params, }; -static int raumfeld_line_suspend(struct platform_device *pdev, pm_message_t state) +static int raumfeld_line_suspend(struct snd_soc_card *card) { raumfeld_enable_audio(false); return 0; } -static int raumfeld_line_resume(struct platform_device *pdev) +static int raumfeld_line_resume(struct snd_soc_card *card) { raumfeld_enable_audio(true); return 0; diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index b222a7d..7b72901 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -226,8 +226,7 @@ static int zylonite_remove(struct platform_device *pdev) return 0; } -static int zylonite_suspend_post(struct platform_device *pdev, - pm_message_t state) +static int zylonite_suspend_post(struct snd_soc_card *card) { if (clk_pout) clk_disable(pout); @@ -235,7 +234,7 @@ static int zylonite_suspend_post(struct platform_device *pdev, return 0; } -static int zylonite_resume_pre(struct platform_device *pdev) +static int zylonite_resume_pre(struct snd_soc_card *card) { int ret = 0; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 14861f9..446838e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1011,7 +1011,7 @@ static int soc_suspend(struct device *dev) } if (card->suspend_pre) - card->suspend_pre(pdev, PMSG_SUSPEND); + card->suspend_pre(card); for (i = 0; i < card->num_rtd; i++) { struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; @@ -1078,7 +1078,7 @@ static int soc_suspend(struct device *dev) } if (card->suspend_post) - card->suspend_post(pdev, PMSG_SUSPEND); + card->suspend_post(card); return 0; } @@ -1090,7 +1090,6 @@ static void soc_resume_deferred(struct work_struct *work) { struct snd_soc_card *card = container_of(work, struct snd_soc_card, deferred_resume_work); - struct platform_device *pdev = to_platform_device(card->dev); struct snd_soc_codec *codec; int i; @@ -1104,7 +1103,7 @@ static void soc_resume_deferred(struct work_struct *work) snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D2); if (card->resume_pre) - card->resume_pre(pdev); + card->resume_pre(card); /* resume AC97 DAIs */ for (i = 0; i < card->num_rtd; i++) { @@ -1179,7 +1178,7 @@ static void soc_resume_deferred(struct work_struct *work) } if (card->resume_post) - card->resume_post(pdev); + card->resume_post(card); dev_dbg(card->dev, "resume work completed\n"); -- 1.7.2.3

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Re: [alsa-devel] [[PATCH] - hdspm 1/2] Add support for RME RayDAT and AIO
by Takashi Iwai 26 Jan '11

26 Jan '11

At Wed, 26 Jan 2011 16:58:29 +0100, Adrian Knoth wrote: > > diff --git a/include/hdspm.h b/include/hdspm.h > index 81990b2..0caf472 100644 > --- a/include/hdspm.h > +++ b/include/hdspm.h ... > struct hdspm_version { > + uint8_t card_type; /* enum hdspm_io_type */ > + char cardname[20]; > + unsigned int serial; > unsigned short firmware_rev; > + int addons; > }; > > #define SNDRV_HDSPM_IOCTL_GET_VERSION _IOR('H', 0x43, struct hdspm_version) Changing the existing ioctl isn't so pretty. If any, I'd vote for changing the ioctl number. > +/* These tables map the ALSA channels 1..N to the channels that we > + need to use in order to find the relevant channel buffer. RME > + refers to this kind of mapping as between "the ADAT channel and > + the DMA channel." We index it using the logical audio channel, > + and the value is the DMA channel (i.e. channel buffer number) > + where the data for that channel can be read/written from/to. > +*/ > + > +static char channel_map_unity_ss[HDSPM_MAX_CHANNELS] = { > + 0, 1, 2, 3, 4, 5, 6, 7, > + 8, 9, 10, 11, 12, 13, 14, 15, > + 16, 17, 18, 19, 20, 21, 22, 23, > + 24, 25, 26, 27, 28, 29, 30, 31, > + 32, 33, 34, 35, 36, 37, 38, 39, > + 40, 41, 42, 43, 44, 45, 46, 47, > + 48, 49, 50, 51, 52, 53, 54, 55, > + 56, 57, 58, 59, 60, 61, 62, 63 > +}; Please don't put static arrays in the public header files. Otherwise it looks OK. It's huge, and maybe there are something to be fixed. But We can fix things after merging. However, the problem is that this patch isn't applicable cleanly to the latest sound git tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git I can change the patch path easily, but the contents get rejected at this time, and I have no gut to fix manually. Could you rebase your patch to the git tree above? Or, you can use alsa-driver-snapshot tarball if you don't want to use the whole kernel git tree. thanks, Takashi

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[alsa-devel] [hdspm: Add RME RayDAT/AIO 0/2] Reworked support for RayDAT/AIO
by Adrian Knoth 26 Jan '11

26 Jan '11

Hi! Finally, here's the patch that adds RME RayDAT and AIO support to hdpsm. Code was taken from Florian Faber's http://wiki.linuxproaudio.org/index.php/Driver:hdspe revision (2010-12-25) and heavily modified to (almost) comply with the coding style. I also enabled S/PDIF input and augmented the code to allow for all period sizes between 64 and 4096 samples on RayDATs/AIOs. As pointed out by Takashi earlier this month, this version now doesn't use typedefs or enums in ioctl structs. Obviously, the patch is extremely long, it adds MMAP access, support for the additional TCO hardware module, more MIDI ports and probably a lot more. Cheers PS: This is the third version of the patch, now without conflicting ioctls (hopefully) and without static arrays in the header file. Adrian Knoth (2): ALSA: Add support for RME RayDAT and AIO ALSA: Add RayDAT and AIO strings to Kconfig include/sound/hdspm.h | 349 ++++- sound/pci/Kconfig | 6 +- sound/pci/rme9652/hdspm.c | 4231 +++++++++++++++++++++++++++++++++------------ 3 files changed, 3427 insertions(+), 1159 deletions(-) -- 1.7.2.3

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[alsa-devel] [PATCH] ASoC: EDB93xx machine sound driver with CS4271
by Alexander Sverdlin 26 Jan '11

26 Jan '11

From: Alexander Sverdlin <subaparts(a)yandex.ru> Added support for EDB93xx sound with CS4271 CODEC. Features: - Playback, Capture - Sample rates from 8kHz to 96kHz tested Limitations: - No DAPM support Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> --- sound/soc/ep93xx/Kconfig | 9 +++ sound/soc/ep93xx/Makefile | 2 + sound/soc/ep93xx/edb93xx.c | 142 ++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 153 insertions(+), 0 deletions(-) diff --git a/sound/soc/ep93xx/Kconfig b/sound/soc/ep93xx/Kconfig index 5742904..91a28de 100644 --- a/sound/soc/ep93xx/Kconfig +++ b/sound/soc/ep93xx/Kconfig @@ -30,3 +30,12 @@ config SND_EP93XX_SOC_SIMONE help Say Y or M here if you want to add support for AC97 audio on the Simplemachines Sim.One board. + +config SND_EP93XX_SOC_EDB93XX + tristate "SoC Audio support for Cirrus Logic EDB93xx boards" + depends on SND_EP93XX_SOC && (MACH_EDB9301 || MACH_EDB9302 || MACH_EDB9302A || MACH_EDB9307A || MACH_EDB9315A) + select SND_EP93XX_SOC_I2S + select SND_SOC_CS4271 + help + Say Y or M here if you want to add support for I2S audio on the + Cirrus Logic EDB93xx boards. diff --git a/sound/soc/ep93xx/Makefile b/sound/soc/ep93xx/Makefile index 8e7977f..5514146 100644 --- a/sound/soc/ep93xx/Makefile +++ b/sound/soc/ep93xx/Makefile @@ -10,6 +10,8 @@ obj-$(CONFIG_SND_EP93XX_SOC_AC97) += snd-soc-ep93xx-ac97.o # EP93XX Machine Support snd-soc-snappercl15-objs := snappercl15.o snd-soc-simone-objs := simone.o +snd-soc-edb93xx-objs := edb93xx.o obj-$(CONFIG_SND_EP93XX_SOC_SNAPPERCL15) += snd-soc-snappercl15.o obj-$(CONFIG_SND_EP93XX_SOC_SIMONE) += snd-soc-simone.o +obj-$(CONFIG_SND_EP93XX_SOC_EDB93XX) += snd-soc-edb93xx.o diff --git a/sound/soc/ep93xx/edb93xx.c b/sound/soc/ep93xx/edb93xx.c new file mode 100644 index 0000000..b270085 --- /dev/null +++ b/sound/soc/ep93xx/edb93xx.c @@ -0,0 +1,142 @@ +/* + * SoC audio for EDB93xx + * + * Copyright (c) 2010 Alexander Sverdlin <subaparts(a)yandex.ru> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * This driver support CS4271 codec being master or slave, working + * in control port mode, connected either via SPI or I2C. + * The data format accepted is I2S or left-justified. + * DAPM support not implemented. + */ + +#include <linux/platform_device.h> +#include <linux/gpio.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include "ep93xx-pcm.h" + +#define edb93xx_has_audio() (machine_is_edb9301() || \ + machine_is_edb9302() || \ + machine_is_edb9302a() || \ + machine_is_edb9307a() || \ + machine_is_edb9315a()) + +static int edb93xx_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + 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; + int err; + unsigned int rate = params_rate(params); + /* + * We set LRCLK equal to `rate' and SCLK = LRCLK * 64, + * because our sample size is 32 bit * 2 channels. + * I2S standard permits us to transmit more bits than + * the codec uses. + * MCLK = SCLK * 4 is the best recommended value, + * but we have to fall back to ratio 2 for higher + * sample rates. + */ + unsigned int mclk_rate = rate * 64 * ((rate <= 48000) ? 4 : 2); + + err = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBS_CFS); + if (err) + return err; + + err = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBS_CFS); + if (err) + return err; + + err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk_rate, + SND_SOC_CLOCK_IN); + if (err) + return err; + + return snd_soc_dai_set_sysclk(cpu_dai, 0, mclk_rate, + SND_SOC_CLOCK_OUT); +} + +static struct snd_soc_ops edb93xx_ops = { + .hw_params = edb93xx_hw_params, +}; + +static struct snd_soc_dai_link edb93xx_dai = { + .name = "CS4271", + .stream_name = "CS4271 HiFi", + .platform_name = "ep93xx-pcm-audio", + .cpu_dai_name = "ep93xx-i2s", + .codec_name = "spi0.0", + .codec_dai_name = "cs4271-hifi", + .ops = &edb93xx_ops, +}; + +static struct snd_soc_card snd_soc_edb93xx = { + .name = "EDB93XX", + .dai_link = &edb93xx_dai, + .num_links = 1, +}; + +static struct platform_device *edb93xx_snd_device; + +static int __init edb93xx_init(void) +{ + int ret; + + if (!edb93xx_has_audio()) + return -ENODEV; + + ret = ep93xx_i2s_acquire(EP93XX_SYSCON_DEVCFG_I2SONAC97, + EP93XX_SYSCON_I2SCLKDIV_ORIDE | + EP93XX_SYSCON_I2SCLKDIV_SPOL); + if (ret) + return ret; + + edb93xx_snd_device = platform_device_alloc("soc-audio", -1); + if (!edb93xx_snd_device) { + ret = -ENOMEM; + goto free_i2s; + } + + platform_set_drvdata(edb93xx_snd_device, &snd_soc_edb93xx); + ret = platform_device_add(edb93xx_snd_device); + if (ret) + goto device_put; + + return 0; + +device_put: + platform_device_put(edb93xx_snd_device); +free_i2s: + ep93xx_i2s_release(); + return ret; +} +module_init(edb93xx_init); + +static void __exit edb93xx_exit(void) +{ + platform_device_unregister(edb93xx_snd_device); + ep93xx_i2s_release(); +} +module_exit(edb93xx_exit); + +MODULE_AUTHOR("Alexander Sverdlin <subaparts(a)yandex.ru>"); +MODULE_DESCRIPTION("ALSA SoC EDB93xx"); +MODULE_LICENSE("GPL");

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