November 2009 - Alsa-devel - mailman.alsa-project.org

[PATCH] ALSA: ASoc: pxa/raumfeld: adopt new snd_soc_dai_set_pll() API
by Daniel Mack 30 Nov '09

30 Nov '09

ALSA's for-2.6.33 branch has a new source argument to snd_soc_dai_set_pll(). Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/pxa/raumfeld.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c index 5d7bb8c..e713181 100644 --- a/sound/soc/pxa/raumfeld.c +++ b/sound/soc/pxa/raumfeld.c @@ -116,7 +116,7 @@ static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream, return ret; /* setup the CPU DAI */ - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, clk); + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, clk); if (ret < 0) return ret; @@ -205,7 +205,7 @@ static int raumfeld_ak4104_hw_params(struct snd_pcm_substream *substream, return ret; /* setup the CPU DAI */ - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, clk); + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, clk); if (ret < 0) return ret; -- 1.6.5.2

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[alsa-devel] [PATCH 2/2] opti-miro: add PnP detection
by Krzysztof Helt 30 Nov '09

30 Nov '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> The PCM12 and PCM20 can be set into the ISA PnP mode. The PCM12 PnP was sold as the PnP device. Add code to handle detection of these cards using ISA PnP framework. Tested on the PCM20 in PnP mode. The PCM12 PnP has the same MS Windows INF file except for a card name displayed for user. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/opti9xx/miro.c | 203 +++++++++++++++++++++++++++++++++++++++++++--- 1 files changed, 192 insertions(+), 11 deletions(-) diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index c67bc3c..6123c75 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -25,6 +25,7 @@ #include <linux/init.h> #include <linux/err.h> #include <linux/isa.h> +#include <linux/pnp.h> #include <linux/delay.h> #include <linux/slab.h> #include <linux/ioport.h> @@ -60,6 +61,9 @@ static int dma1 = SNDRV_DEFAULT_DMA1; /* 0,1,3 */ static int dma2 = SNDRV_DEFAULT_DMA1; /* 0,1,3 */ static int wss; static int ide; +#ifdef CONFIG_PNP +static int isapnp = 1; /* Enable ISA PnP detection */ +#endif module_param(index, int, 0444); MODULE_PARM_DESC(index, "Index value for miro soundcard."); @@ -83,6 +87,10 @@ module_param(wss, int, 0444); MODULE_PARM_DESC(wss, "wss mode"); module_param(ide, int, 0444); MODULE_PARM_DESC(ide, "enable ide port"); +#ifdef CONFIG_PNP +module_param(isapnp, bool, 0444); +MODULE_PARM_DESC(isapnp, "Enable ISA PnP detection for specified soundcard."); +#endif #define OPTi9XX_HW_DETECT 0 #define OPTi9XX_HW_82C928 1 @@ -131,6 +139,21 @@ static char * snd_opti9xx_names[] = { "82C930", "82C931", "82C933" }; +static int snd_miro_pnp_is_probed; + +#ifdef CONFIG_PNP + +static struct pnp_card_device_id snd_miro_pnpids[] = { + /* PCM20 and PCM12 in PnP mode */ + { .id = "MIR0924", + .devs = { { "MIR0000" }, { "MIR0002" }, { "MIR0005" } }, }, + { .id = "" } +}; + +MODULE_DEVICE_TABLE(pnp_card, snd_miro_pnpids); + +#endif /* CONFIG_PNP */ + /* * ACI control */ @@ -781,17 +804,23 @@ static int __devinit snd_miro_init(struct snd_miro *chip, chip->mpu_port = -1; chip->mpu_irq = -1; + chip->pwd_reg = 3; + +#ifdef CONFIG_PNP + if (isapnp && chip->mc_base) + /* PnP resource gives the least 10 bits */ + chip->mc_base |= 0xc00; + else +#endif + chip->mc_base = 0xf8c; + switch (hardware) { case OPTi9XX_HW_82C929: - chip->mc_base = 0xf8c; chip->password = 0xe3; - chip->pwd_reg = 3; break; case OPTi9XX_HW_82C924: - chip->mc_base = 0xf8c; chip->password = 0xe5; - chip->pwd_reg = 3; break; default: @@ -1014,17 +1043,22 @@ static int __devinit snd_miro_configure(struct snd_miro *chip) return -EINVAL; } - switch (chip->wss_base) { - case 0x530: + /* PnP resource says it decodes only 10 bits of address */ + switch (chip->wss_base & 0x3ff) { + case 0x130: + chip->wss_base = 0x530; wss_base_bits = 0x00; break; - case 0x604: + case 0x204: + chip->wss_base = 0x604; wss_base_bits = 0x03; break; - case 0xe80: + case 0x280: + chip->wss_base = 0xe80; wss_base_bits = 0x01; break; - case 0xf40: + case 0x340: + chip->wss_base = 0xf40; wss_base_bits = 0x02; break; default: @@ -1238,7 +1272,7 @@ static int __devinit snd_card_miro_aci_detect(struct snd_card *card, static void snd_card_miro_free(struct snd_card *card) { struct snd_miro *miro = card->private_data; - + release_and_free_resource(miro->res_aci_port); if (miro->aci) miro->aci->aci_port = 0; @@ -1370,6 +1404,12 @@ static int __devinit snd_miro_probe(struct snd_card *card) static int __devinit snd_miro_isa_match(struct device *devptr, unsigned int n) { +#ifdef CONFIG_PNP + if (snd_miro_pnp_is_probed) + return 0; + if (isapnp) + return 0; +#endif return 1; } @@ -1487,14 +1527,155 @@ static struct isa_driver snd_miro_driver = { }, }; +#ifdef CONFIG_PNP + +static int __devinit snd_card_miro_pnp(struct snd_miro *chip, + struct pnp_card_link *card, + const struct pnp_card_device_id *pid) +{ + struct pnp_dev *pdev; + int err; + struct pnp_dev *devmpu; + struct pnp_dev *devmc; + + pdev = pnp_request_card_device(card, pid->devs[0].id, NULL); + if (pdev == NULL) + return -EBUSY; + + devmpu = pnp_request_card_device(card, pid->devs[1].id, NULL); + if (devmpu == NULL) + return -EBUSY; + + devmc = pnp_request_card_device(card, pid->devs[2].id, NULL); + if (devmc == NULL) + return -EBUSY; + + err = pnp_activate_dev(pdev); + if (err < 0) { + snd_printk(KERN_ERR "AUDIO pnp configure failure: %d\n", err); + return err; + } + + err = pnp_activate_dev(devmc); + if (err < 0) { + snd_printk(KERN_ERR "OPL syntg pnp configure failure: %d\n", + err); + return err; + } + + port = pnp_port_start(pdev, 1); + fm_port = pnp_port_start(pdev, 2) + 8; + + /* + * The MC(0) is never accessed and the miroSOUND PCM20 card does not + * include it in the PnP resource range. OPTI93x include it. + */ + chip->mc_base = pnp_port_start(devmc, 0) - 1; + chip->mc_base_size = pnp_port_len(devmc, 0) + 1; + + irq = pnp_irq(pdev, 0); + dma1 = pnp_dma(pdev, 0); + dma2 = pnp_dma(pdev, 1); + + if (mpu_port > 0) { + err = pnp_activate_dev(devmpu); + if (err < 0) { + snd_printk(KERN_ERR "MPU401 pnp configure failure\n"); + mpu_port = -1; + return err; + } + mpu_port = pnp_port_start(devmpu, 0); + mpu_irq = pnp_irq(devmpu, 0); + } + return 0; +} + +static int __devinit snd_miro_pnp_probe(struct pnp_card_link *pcard, + const struct pnp_card_device_id *pid) +{ + struct snd_card *card; + int err; + struct snd_miro *miro; + + if (snd_miro_pnp_is_probed) + return -EBUSY; + if (!isapnp) + return -ENODEV; + err = snd_card_create(index, id, THIS_MODULE, + sizeof(struct snd_miro), &card); + if (err < 0) + return err; + + card->private_free = snd_card_miro_free; + miro = card->private_data; + + err = snd_card_miro_pnp(miro, pcard, pid); + if (err) { + snd_card_free(card); + return err; + } + + /* only miroSOUND PCM20 and PCM12 == OPTi924 */ + err = snd_miro_init(miro, OPTi9XX_HW_82C924); + if (err) { + snd_card_free(card); + return err; + } + + err = snd_miro_opti_check(miro); + if (err) { + snd_printk(KERN_ERR "OPTI chip not found\n"); + snd_card_free(card); + return err; + } + + snd_card_set_dev(card, &pcard->card->dev); + err = snd_miro_probe(card); + if (err < 0) { + snd_card_free(card); + return err; + } + pnp_set_card_drvdata(pcard, card); + snd_miro_pnp_is_probed = 1; + return 0; +} + +static void __devexit snd_miro_pnp_remove(struct pnp_card_link * pcard) +{ + snd_card_free(pnp_get_card_drvdata(pcard)); + pnp_set_card_drvdata(pcard, NULL); + snd_miro_pnp_is_probed = 0; +} + +static struct pnp_card_driver miro_pnpc_driver = { + .flags = PNP_DRIVER_RES_DISABLE, + .name = "miro", + .id_table = snd_miro_pnpids, + .probe = snd_miro_pnp_probe, + .remove = __devexit_p(snd_miro_pnp_remove), +}; +#endif + static int __init alsa_card_miro_init(void) { +#ifdef CONFIG_PNP + pnp_register_card_driver(&miro_pnpc_driver); + if (snd_miro_pnp_is_probed) + return 0; + pnp_unregister_card_driver(&miro_pnpc_driver); +#endif return isa_register_driver(&snd_miro_driver, 1); } static void __exit alsa_card_miro_exit(void) { - isa_unregister_driver(&snd_miro_driver); + if (!snd_miro_pnp_is_probed) { + isa_unregister_driver(&snd_miro_driver); + return; + } +#ifdef CONFIG_PNP + pnp_unregister_card_driver(&miro_pnpc_driver); +#endif } module_init(alsa_card_miro_init) -- 1.6.4 ---------------------------------------------------------------------- Konkurs dla fanow Hannah Montana! 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[alsa-devel] [PATCH 1/2] ALSA: AACI: fix AC97 multiple-open bug
by Russell King - ARM Linux 29 Nov '09

29 Nov '09

Signed-off-by: Russell King <rmk+kernel(a)arm.linux.org.uk> --- sound/arm/aaci.c | 4 ++++ 1 files changed, 4 insertions(+), 0 deletions(-) diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index a03fe80..1456fc7 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -501,6 +501,10 @@ static int aaci_pcm_hw_params(struct snd_pcm_substream *substream, int err; aaci_pcm_hw_free(substream); + if (aacirun->pcm_open) { + snd_ac97_pcm_close(aacirun->pcm); + aacirun->pcm_open = 0; + } err = devdma_hw_alloc(NULL, substream, params_buffer_bytes(params)); -- 1.6.2.5

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[alsa-devel] i.MX audio support
by Sascha Hauer 28 Nov '09

28 Nov '09

Hi All, This series adds support for AOSC on i.MX platforms. It uses DMA where available (i.MX27) and fiq otherwise. The driver works on all Phytec i.MX platforms, so the i.MX27, i.MX31 and the i.MX35 are supported including AC97 support. There already is a driver for i.MX in the tree, this one is a nearly complete rewrite. I haven't touched the in Kernel driver with these patches. The in Kernel driver has several problems, it doesn't use ioremap, uses direct pointer derefs instead of proper access functions and the only board supported so far is itself not supported in mainline. AC97 support is a problem on i.MX. I found no way to tell the SSI controller which slots it should capture, so all slots the codec decides to send go into one single FIFO. The WM9712 I worked with sends data in slot 12 which we have to sort out manually from the FIFO, so I think there is no way to work around the fiq handler using the DMA engine with AC97. The devdma code is actually a copy of sound/arm/devdma.[ch]. It should probably be selected somehow instead of duplicated. I'm not very familiar with ALSA, so I expect some comments in this area. Let me know what has to be done to get this into mainline. This series depends on the SSI resources and audmux support being present. I posted these patches on arm linux kernel recently and they are available here: git://git.pengutronix.de/git/imx/linux-2.6.git mxc-master Any comments welcome. Sascha The following changes since commit 3f92a8bd5fb13e7e2505c65d1548910eaa843024: Uwe Kleine-König (1): imx: copy constants from mx3x.h to mx35.h using the appropriate namespace are available in the git repository at: git://git.pengutronix.de/git/imx/linux-2.6.git asoc Sascha Hauer (12): mx3: Add SSI pins to iomux table mxc: iomux v3: remove resource handling add a mc13783 codec driver i.MX31 clock: rename SSI clocks to driver name mxc: mx1/mx2 DMA: add a possibility to create an endless DMA transfer imx-ssi sound driver add phycore sound support sound/soc/imx: Makefile/Kconfig changes for new driver pcm038: add sound support pcm043: add sound support pca100: add sound support pcm037: Add sound support arch/arm/mach-mx2/pca100.c | 52 + arch/arm/mach-mx2/pcm038.c | 23 +- arch/arm/mach-mx3/clock.c | 4 +- arch/arm/mach-mx3/pcm037.c | 72 ++ arch/arm/mach-mx3/pcm043.c | 102 ++ arch/arm/plat-mxc/Makefile | 1 + arch/arm/plat-mxc/dma-mx1-mx2.c | 3 +- arch/arm/plat-mxc/include/mach/dma-mx1-mx2.h | 8 + arch/arm/plat-mxc/include/mach/iomux-mx3.h | 8 + arch/arm/plat-mxc/include/mach/iomux-v3.h | 17 +- arch/arm/plat-mxc/include/mach/ssi.h | 17 + arch/arm/plat-mxc/iomux-v3.c | 36 +- arch/arm/plat-mxc/ssi-fiq.S | 134 +++ sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/mc13783.c | 739 ++++++++++++++ sound/soc/codecs/mc13783.h | 29 + sound/soc/imx/Kconfig | 15 +- sound/soc/imx/Makefile | 9 +- sound/soc/imx/devdma.c | 80 ++ sound/soc/imx/devdma.h | 3 + sound/soc/imx/imx-ssi.c | 1349 ++++++++++++++++++++++++++ sound/soc/imx/imx-ssi.h | 24 + sound/soc/imx/phycore.c | 181 ++++ 24 files changed, 2851 insertions(+), 60 deletions(-) create mode 100644 arch/arm/plat-mxc/include/mach/ssi.h create mode 100644 arch/arm/plat-mxc/ssi-fiq.S create mode 100644 sound/soc/codecs/mc13783.c create mode 100644 sound/soc/codecs/mc13783.h create mode 100644 sound/soc/imx/devdma.c create mode 100644 sound/soc/imx/devdma.h create mode 100644 sound/soc/imx/imx-ssi.c create mode 100644 sound/soc/imx/imx-ssi.h create mode 100644 sound/soc/imx/phycore.c

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Re: [alsa-devel] [Alsa-user] m-audio delta 1010 inputs crackle at 96 kHz sample rate with all newer kernels
by Clemens Ladisch 27 Nov '09

27 Nov '09

francis keyes wrote: > As soon as I switch to 96kHz in envy24control, I can hear lots of > crackle and pop noises through the inputs that are also recorded when > I try to record. I first noticed this after installing a new kernel. Which kernel? Best regards, Clemens

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[alsa-devel] [PATCH] ASoC: Implement initial S3C64XX IISv4 support
by Mark Brown 27 Nov '09

27 Nov '09

The IISv4 port can be controlled by the IISv2 driver so copy over that with light modifications, including use of platform data to get the pin configuration. Currently this is S3C6410 specific due to the DMA configuration - in the future when we have IISv4 specific support we may want to split out a generic IISv4 portion. Since this allows the SMDK64xx machine driver to work update it to reflect the API changes involved and mark it as not BROKEN. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- This depends on the arch/arm changes I posted to linux-arm-kernel a minute ago. I'll not push it until it looks like those will be merged. sound/soc/s3c24xx/Kconfig | 10 +++++++--- sound/soc/s3c24xx/Makefile | 2 ++ sound/soc/s3c24xx/s3c64xx-i2s.h | 1 + sound/soc/s3c24xx/smdk64xx_wm8580.c | 6 ++---- 4 files changed, 12 insertions(+), 7 deletions(-) diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index b489f1a..1b98850 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -24,6 +24,11 @@ config SND_S3C64XX_SOC_I2S select SND_S3C_I2SV2_SOC select S3C64XX_DMA +config SND_S3C64XX_SOC_I2S_V4 + tristate + select SND_S3C_I2SV2_SOC + select S3C64XX_DMA + config SND_S3C_SOC_PCM tristate @@ -62,11 +67,10 @@ config SND_S3C24XX_SOC_JIVE_WM8750 config SND_S3C64XX_SOC_WM8580 tristate "SoC I2S Audio support for WM8580 on SMDK64XX" depends on SND_S3C24XX_SOC && (MACH_SMDK6400 || MACH_SMDK6410) - depends on BROKEN select SND_SOC_WM8580 - select SND_S3C64XX_SOC_I2S + select SND_S3C64XX_SOC_I2S_V4 help - Sat Y if you want to add support for SoC audio on the SMDK64XX. + Say Y if you want to add support for SoC audio on the SMDK64XX. config SND_S3C24XX_SOC_SMDK2443_WM9710 tristate "SoC AC97 Audio support for SMDK2443 - WM9710" diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index b744657..114028c 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -3,6 +3,7 @@ snd-soc-s3c24xx-objs := s3c-dma.o snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o snd-soc-s3c64xx-i2s-objs := s3c64xx-i2s.o +snd-soc-s3c64xx-i2s-v4-objs := s3c64xx-i2s-v4.o snd-soc-s3c2443-ac97-objs := s3c2443-ac97.o snd-soc-s3c-i2s-v2-objs := s3c-i2s-v2.o snd-soc-s3c-pcm-objs := s3c-pcm.o @@ -12,6 +13,7 @@ obj-$(CONFIG_SND_S3C24XX_SOC_I2S) += snd-soc-s3c24xx-i2s.o obj-$(CONFIG_SND_S3C2443_SOC_AC97) += snd-soc-s3c2443-ac97.o obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o obj-$(CONFIG_SND_S3C64XX_SOC_I2S) += snd-soc-s3c64xx-i2s.o +obj-$(CONFIG_SND_S3C64XX_SOC_I2S_V4) += snd-soc-s3c64xx-i2s-v4.o obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o obj-$(CONFIG_SND_S3C_SOC_PCM) += snd-soc-s3c-pcm.o diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.h b/sound/soc/s3c24xx/s3c64xx-i2s.h index abe7253..a50b2dc 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.h +++ b/sound/soc/s3c24xx/s3c64xx-i2s.h @@ -28,6 +28,7 @@ struct clk; #define S3C64XX_CLKSRC_CDCLK (2) extern struct snd_soc_dai s3c64xx_i2s_dai[]; +extern struct snd_soc_dai s3c64xx_i2s_v4_dai[]; extern struct clk *s3c64xx_i2s_get_clock(struct snd_soc_dai *dai); diff --git a/sound/soc/s3c24xx/smdk64xx_wm8580.c b/sound/soc/s3c24xx/smdk64xx_wm8580.c index efe4901..b5bb129 100644 --- a/sound/soc/s3c24xx/smdk64xx_wm8580.c +++ b/sound/soc/s3c24xx/smdk64xx_wm8580.c @@ -22,8 +22,6 @@ #include "s3c-dma.h" #include "s3c64xx-i2s.h" -#define S3C64XX_I2S_V4 2 - /* SMDK64XX has a 12MHZ crystal attached to WM8580 */ #define SMDK64XX_WM8580_FREQ 12000000 @@ -215,7 +213,7 @@ static struct snd_soc_dai_link smdk64xx_dai[] = { { /* Primary Playback i/f */ .name = "WM8580 PAIF RX", .stream_name = "Playback", - .cpu_dai = &s3c64xx_i2s_dai[S3C64XX_I2S_V4], + .cpu_dai = &s3c64xx_i2s_v4_dai[0], .codec_dai = &wm8580_dai[WM8580_DAI_PAIFRX], .init = smdk64xx_wm8580_init_paifrx, .ops = &smdk64xx_ops, @@ -223,7 +221,7 @@ static struct snd_soc_dai_link smdk64xx_dai[] = { { /* Primary Capture i/f */ .name = "WM8580 PAIF TX", .stream_name = "Capture", - .cpu_dai = &s3c64xx_i2s_dai[S3C64XX_I2S_V4], + .cpu_dai = &s3c64xx_i2s_v4_dai[0], .codec_dai = &wm8580_dai[WM8580_DAI_PAIFTX], .init = smdk64xx_wm8580_init_paiftx, .ops = &smdk64xx_ops, -- 1.6.5.3

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[alsa-devel] [PATCH 2/3] ALSA: ARM: add Raumfeld audio support
by Daniel Mack 27 Nov '09

27 Nov '09

Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: alsa-devel(a)alsa-project.org --- sound/soc/pxa/Kconfig | 9 ++ sound/soc/pxa/Makefile | 2 + sound/soc/pxa/raumfeld.c | 335 ++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 346 insertions(+), 0 deletions(-) create mode 100644 sound/soc/pxa/raumfeld.c diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index dcb3181..9c3bfd9 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -117,6 +117,15 @@ config SND_SOC_ZYLONITE Say Y if you want to add support for SoC audio on the Marvell Zylonite reference platform. +config SND_SOC_RAUMFELD + tristate "SoC Audio support Raumfeld audio adapter" + depends on SND_PXA2XX_SOC && (MACH_RAUMFELD_SPEAKER || MACH_RAUMFELD_CONNECTOR) + select SND_PXA_SOC_SSP + select SND_SOC_CS4270 + select SND_SOC_AK4104 + help + Say Y if you want to add support for SoC audio on Raumfeld devices + config SND_PXA2XX_SOC_MAGICIAN tristate "SoC Audio support for HTC Magician" depends on SND_PXA2XX_SOC && MACH_MAGICIAN diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 6e096b4..f3e08fd 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -23,6 +23,7 @@ snd-soc-zylonite-objs := zylonite.o snd-soc-magician-objs := magician.o snd-soc-mioa701-objs := mioa701_wm9713.o snd-soc-imote2-objs := imote2.o +snd-soc-raumfeld-objs := raumfeld.o obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o @@ -37,3 +38,4 @@ obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o +obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c new file mode 100644 index 0000000..f272269 --- /dev/null +++ b/sound/soc/pxa/raumfeld.c @@ -0,0 +1,335 @@ +/* + * raumfeld_audio.c -- SoC audio for Raumfeld audio devices + * + * Copyright (c) 2009 Daniel Mack <daniel(a)caiaq.de> + * + * based on code from: + * + * Wolfson Microelectronics PLC. + * Openedhand Ltd. + * Liam Girdwood <lrg(a)slimlogic.co.uk> + * Richard Purdie <richard(a)openedhand.com> + * + * 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. + */ + +#include <linux/module.h> +#include <linux/i2c.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> + +#include "../codecs/cs4270.h" +#include "../codecs/ak4104.h" +#include "pxa2xx-pcm.h" +#include "pxa-ssp.h" + +#define GPIO_SPDIF_RESET (38) +#define GPIO_MCLK_RESET (111) +#define GPIO_CODEC_RESET (120) + +static struct i2c_client *max9486_client; +static struct i2c_board_info max9486_hwmon_info = { + I2C_BOARD_INFO("max9485", 0x63), +}; + +#define MAX9485_MCLK_FREQ_112896 0x22 +#define MAX9485_MCLK_FREQ_122880 0x23 + +static void set_max9485_clk(char clk) +{ + i2c_master_send(max9486_client, &clk, 1); +} + +static void raumfeld_enable_audio(bool en) +{ + if (en) { + gpio_set_value(GPIO_MCLK_RESET, 1); + + /* wait some time to let the clocks become stable */ + msleep(100); + + gpio_set_value(GPIO_SPDIF_RESET, 1); + gpio_set_value(GPIO_CODEC_RESET, 1); + } else { + gpio_set_value(GPIO_MCLK_RESET, 0); + gpio_set_value(GPIO_SPDIF_RESET, 0); + gpio_set_value(GPIO_CODEC_RESET, 0); + } +} + +/* CS4270 */ +static int raumfeld_cs4270_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + set_max9485_clk(MAX9485_MCLK_FREQ_112896); + + return snd_soc_dai_set_sysclk(codec_dai, 0, 11289600, 0); +} + +static int raumfeld_cs4270_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->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int fmt, clk = 0; + int ret = 0; + + switch (params_rate(params)) { + case 8000: + case 16000: + case 48000: + case 96000: + set_max9485_clk(MAX9485_MCLK_FREQ_122880); + clk = 12288000; + break; + case 11025: + case 22050: + case 44100: + case 88200: + set_max9485_clk(MAX9485_MCLK_FREQ_112896); + clk = 11289600; + break; + } + + fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS; + + /* setup the CODEC DAI */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, 0); + if (ret < 0) + return ret; + + /* setup the CPU DAI */ + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, clk); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, 0, 1); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops raumfeld_cs4270_ops = { + .startup = raumfeld_cs4270_startup, + .hw_params = raumfeld_cs4270_hw_params, +}; + +static int raumfeld_line_suspend(struct platform_device *pdev, pm_message_t state) +{ + raumfeld_enable_audio(false); + return 0; +} + +static int raumfeld_line_resume(struct platform_device *pdev) +{ + raumfeld_enable_audio(true); + return 0; +} + +static struct snd_soc_dai_link raumfeld_line_dai = { + .name = "CS4270", + .stream_name = "CS4270", + .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP1], + .codec_dai = &cs4270_dai, + .ops = &raumfeld_cs4270_ops, +}; + +static struct snd_soc_card snd_soc_line_raumfeld = { + .name = "Raumfeld analog", + .platform = &pxa2xx_soc_platform, + .dai_link = &raumfeld_line_dai, + .suspend_post = raumfeld_line_suspend, + .resume_pre = raumfeld_line_resume, + .num_links = 1, +}; + + +/* AK4104 */ + +static int raumfeld_ak4104_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->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int fmt, ret = 0, clk = 0; + + switch (params_rate(params)) { + case 8000: + case 16000: + case 48000: + case 96000: + set_max9485_clk(MAX9485_MCLK_FREQ_122880); + clk = 12288000; + break; + case 11025: + case 22050: + case 44100: + case 88200: + set_max9485_clk(MAX9485_MCLK_FREQ_112896); + clk = 11289600; + break; + } + + fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF; + + /* setup the CODEC DAI */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* setup the CPU DAI */ + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, clk); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, 0, 1); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops raumfeld_ak4104_ops = { + .hw_params = raumfeld_ak4104_hw_params, +}; + +static struct snd_soc_dai_link raumfeld_spdif_dai = { + .name = "ak4104", + .stream_name = "Playback", + .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP2], + .codec_dai = &ak4104_dai, + .ops = &raumfeld_ak4104_ops, +}; + +static struct snd_soc_card snd_soc_spdif_raumfeld = { + .name = "Raumfeld S/PDIF", + .platform = &pxa2xx_soc_platform, + .dai_link = &raumfeld_spdif_dai, + .num_links = 1 +}; + +/* raumfeld_audio audio subsystem */ +static struct snd_soc_device raumfeld_line_devdata = { + .card = &snd_soc_line_raumfeld, + .codec_dev = &soc_codec_device_cs4270, +}; + +static struct snd_soc_device raumfeld_spdif_devdata = { + .card = &snd_soc_spdif_raumfeld, + .codec_dev = &soc_codec_device_ak4104, +}; + +static struct platform_device *raumfeld_audio_line_device; +static struct platform_device *raumfeld_audio_spdif_device; + +static int __init raumfeld_audio_init(void) +{ + int ret; + + if (!machine_is_raumfeld_speaker() && + !machine_is_raumfeld_connector()) + return 0; + + max9486_client = i2c_new_device(i2c_get_adapter(0), + &max9486_hwmon_info); + + if (!max9486_client) + return -ENOMEM; + + set_max9485_clk(MAX9485_MCLK_FREQ_122880); + + /* LINE */ + raumfeld_audio_line_device = platform_device_alloc("soc-audio", 0); + if (!raumfeld_audio_line_device) + return -ENOMEM; + + platform_set_drvdata(raumfeld_audio_line_device, + &raumfeld_line_devdata); + raumfeld_line_devdata.dev = &raumfeld_audio_line_device->dev; + ret = platform_device_add(raumfeld_audio_line_device); + if (ret) + platform_device_put(raumfeld_audio_line_device); + + /* no S/PDIF on Speakers */ + if (machine_is_raumfeld_speaker()) + return ret; + + /* S/PDIF */ + raumfeld_audio_spdif_device = platform_device_alloc("soc-audio", 1); + if (!raumfeld_audio_spdif_device) { + platform_device_put(raumfeld_audio_line_device); + return -ENOMEM; + } + + platform_set_drvdata(raumfeld_audio_spdif_device, + &raumfeld_spdif_devdata); + raumfeld_spdif_devdata.dev = &raumfeld_audio_spdif_device->dev; + ret = platform_device_add(raumfeld_audio_spdif_device); + if (ret) { + platform_device_put(raumfeld_audio_line_device); + platform_device_put(raumfeld_audio_spdif_device); + } + + raumfeld_enable_audio(true); + + return ret; +} + +static void __exit raumfeld_audio_exit(void) +{ + raumfeld_enable_audio(false); + + platform_device_unregister(raumfeld_audio_line_device); + + if (machine_is_raumfeld_connector()) + platform_device_unregister(raumfeld_audio_spdif_device); + + i2c_unregister_device(max9486_client); + + gpio_free(GPIO_MCLK_RESET); + gpio_free(GPIO_CODEC_RESET); + gpio_free(GPIO_SPDIF_RESET); +} + +module_init(raumfeld_audio_init); +module_exit(raumfeld_audio_exit); + +/* Module information */ +MODULE_AUTHOR("Daniel Mack <daniel(a)caiaq.de>"); +MODULE_DESCRIPTION("Raumfeld audio SoC"); +MODULE_LICENSE("GPL"); -- 1.6.5.2

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[alsa-devel] Sound support for Neuros OSD2 Davinci based board.
by A. Porodko 27 Nov '09

27 Nov '09

Hello, Here is a patch for Neuros OSD2 Davinci (DM6446) based board sound support. Patch made against 2.6.32-rc6 kernel. ------------------------------------------ -- Best regards Andrey A. Porodko

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[alsa-devel] [PATCH - alsa-plugins 1/2] a52 - fix 5.1 channel order with recent libavcodec
by Anssi Hannula 27 Nov '09

27 Nov '09

As of SVN r18540 libavcodec expects 5.1 channel audio with SMPTE channel order. Fix ALSA a52 plugin to use that order when built against such a libavcodec. Minor version of libavcodec was raised on the same day (Apr 17th 2009), so use that for the check. Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> --- a52/pcm_a52.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/a52/pcm_a52.c b/a52/pcm_a52.c index 1abfdaa..b95d7d4 100644 --- a/a52/pcm_a52.c +++ b/a52/pcm_a52.c @@ -170,7 +170,13 @@ static int fill_data(snd_pcm_ioplug_t *io, static unsigned int ch_index[3][6] = { { 0, 1 }, { 0, 1, 2, 3 }, +#if LIBAVCODEC_VERSION_MAJOR > 52 || (LIBAVCODEC_VERSION_MAJOR == 52 && LIBAVCODEC_VERSION_MINOR >= 26) + /* current libavcodec expects SMPTE order */ + { 0, 1, 4, 5, 2, 3 }, +#else + /* libavcodec older than r18540 expects A52 order */ { 0, 4, 1, 2, 3, 5 }, +#endif }; /* flatten copy to n-channel interleaved */ dst_step = io->channels; -- 1.6.4.4

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[alsa-devel] ASoC: Revert missing reset_err in wm97*.c
by Takashi Iwai 27 Nov '09

27 Nov '09

FYI, I committed the fix below to topic/asoc branch now. thanks, Takashi === From a22eaf4ce106404f6c5283da30b4d514ede964c1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai <tiwai(a)suse.de> Date: Fri, 27 Nov 2009 15:14:09 +0100 Subject: [PATCH] ASoC: Revert missing reset_err in wm97*.c The commit fe3e78e073d25308756f38019956061153267769 ASoC: Factor out snd_soc_init_card() removed the error paths that are still valid for wm97* codecs, causing the compile errors like sound/soc/codecs/wm9705.c:399: error: label 'reset_err' used but not defined sound/soc/codecs/wm9712.c:687: error: label 'reset_err' used but not defined sound/soc/codecs/wm9713.c:1237: error: label 'reset_err' used but not defined Revert the removed error path codes. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/codecs/wm9705.c | 2 ++ sound/soc/codecs/wm9712.c | 2 ++ sound/soc/codecs/wm9713.c | 2 ++ 3 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index dfffc6c..ec54c6d 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -404,6 +404,8 @@ static int wm9705_soc_probe(struct platform_device *pdev) return 0; +reset_err: + snd_soc_free_pcms(socdev); pcm_err: snd_soc_free_ac97_codec(codec); codec_err: diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 2a08722..0ac1215 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -697,6 +697,8 @@ static int wm9712_soc_probe(struct platform_device *pdev) return 0; +reset_err: + snd_soc_free_pcms(socdev); pcm_err: snd_soc_free_ac97_codec(codec); diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 00bac31..4d74ecb 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1249,6 +1249,8 @@ static int wm9713_soc_probe(struct platform_device *pdev) return 0; +reset_err: + snd_soc_free_pcms(socdev); pcm_err: snd_soc_free_ac97_codec(codec); -- 1.6.5.3

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