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

[alsa-devel] [PATCH 1/1] hdsp: allow firmware loading from inside the kernel
by Raphaël Doursenaud 12 May '09

12 May '09

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 From: Raphaël Doursenaud <rdoursenaud(a)free.fr> Allow the use of the FIRMWARE_IN_KERNEL option with hdsp cards and in-kernel driver. Also corrected a typo in the comment. Signed-off-by: Raphael Doursenaud <rdoursenaud(a)free.fr> - --- sound/pci/rme9652/hdsp.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 314e735..d0d4629 100644 - --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -402,9 +402,9 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin"); #define HDSP_DMA_AREA_BYTES ((HDSP_MAX_CHANNELS+1) * HDSP_CHANNEL_BUFFER_BYTES) #define HDSP_DMA_AREA_KILOBYTES (HDSP_DMA_AREA_BYTES/1024) - -/* use hotplug firmeare loader? */ +/* use hotplug firmware loader? */ #if defined(CONFIG_FW_LOADER) || defined(CONFIG_FW_LOADER_MODULE) - -#if !defined(HDSP_USE_HWDEP_LOADER) && !defined(CONFIG_SND_HDSP) +#if !defined(HDSP_USE_HWDEP_LOADER) #define HDSP_FW_LOADER #endif #endif - -- Raphaël Doursenaud -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAkoJFCQACgkQaZKmNAdXaVVliACaA5BMknMAXgOEN/7YQmWGPwUh uRgAn0BiIwAtGU/AOi+QRn81k8apvHfi =+4Kw -----END PGP SIGNATURE-----

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[alsa-devel] [PATCH] soc: remove driver_data direct access of struct device
by Roel Kluin 11 May '09

11 May '09

diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 510efa6..e4547de 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -1473,8 +1473,8 @@ static int wm8400_codec_probe(struct platform_device *dev) codec = &priv->codec; codec->private_data = priv; - codec->control_data = dev->dev.driver_data; - priv->wm8400 = dev->dev.driver_data; + codec->control_data = dev_get_drvdata(&dev->dev); + priv->wm8400 = dev_get_drvdata(&dev->dev); ret = regulator_bulk_get(priv->wm8400->dev, ARRAY_SIZE(power), &power[0]); diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index e043e3f..7a20587 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -666,14 +666,14 @@ static int __devinit wm8731_spi_probe(struct spi_device *spi) codec->hw_write = (hw_write_t)wm8731_spi_write; codec->dev = &spi->dev; - spi->dev.driver_data = wm8731; + dev_set_drvdata(&spi->dev, wm8731); return wm8731_register(wm8731); } static int __devexit wm8731_spi_remove(struct spi_device *spi) { - struct wm8731_priv *wm8731 = spi->dev.driver_data; + struct wm8731_priv *wm8731 = dev_get_drvdata(&spi->dev); wm8731_unregister(wm8731); diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index a6e8f3f..d121e58 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1822,14 +1822,14 @@ static int __devinit wm8753_spi_probe(struct spi_device *spi) codec->hw_write = (hw_write_t)wm8753_spi_write; codec->dev = &spi->dev; - spi->dev.driver_data = wm8753; + dev_set_drvdata(&spi->dev, wm8753); return wm8753_register(wm8753); } static int __devexit wm8753_spi_remove(struct spi_device *spi) { - struct wm8753_priv *wm8753 = spi->dev.driver_data; + struct wm8753_priv *wm8753 = dev_get_drvdata(&spi->dev); wm8753_unregister(wm8753); return 0; }

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[alsa-devel] MacbookPro5,1 & hda-analyzer
by Ozan Çağlayan 11 May '09

11 May '09

Hi, I just bought a MacbookPro5,1 and having difficulties with the onboard HDA chipset. * No sound can be heard of it if model= is not set to mbp3: [ 12.009422] hda_codec: Unknown model for ALC883, trying auto-probe from BIOS... http://www.alsa-project.org/db/?f=4b2f257e72523bccafd3ace1d3c7044fb5f09ff8 * If model=mbp3 is given, it's possible to get sound from it but with many annoying problems: - You have to switch to 6ch mode for getting sound, doesn't work with 2ch mode, - The sound quality is very very bad, - Once you switch to 6ch mode, - You have to increment the volume of *Line-in* to have hearable sound from *internal speakers*, - You have to increment the volume of *Front* to have hearable sound from *headphones*, - The two controls are not mutually exclusive, you have to mute one of them, if you're using the other. And then I decided to play with hda-analyzer. After a bit of playing, I started to get sound from the board without the model=mbp3. But the mixers are still behaving somewhat strange. Before actually reporting any feedback on what I've switched and did, what do you suggest me to do? Is setting the model= to something, something similar to what I was doing with hda-analyzer? Thanks Ozan Caglayan

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[alsa-devel] [PATCH] ASoC: em-x270: make the driver support also eXeda and CM-X300 machines
by Mike Rapoport 11 May '09

11 May '09

Signed-off-by: Mike Rapoport <mike(a)compulab.co.il> --- sound/soc/pxa/Kconfig | 4 ++-- sound/soc/pxa/em-x270.c | 9 +++++---- 2 files changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index ad8a10f..f8cc974 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -89,13 +89,13 @@ config SND_PXA2XX_SOC_E800 Toshiba e800 PDA config SND_PXA2XX_SOC_EM_X270 - tristate "SoC Audio support for CompuLab EM-x270" + tristate "SoC Audio support for CompuLab EM-x270, eXeda and CM-X300" depends on SND_PXA2XX_SOC && MACH_EM_X270 select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help Say Y if you want to add support for SoC audio on - CompuLab EM-x270. + CompuLab EM-x270, eXeda and CM-X300 machines. config SND_PXA2XX_SOC_PALM27X bool "SoC Audio support for Palm T|X, T5 and LifeDrive" diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c index 949be9c..f4756e4 100644 --- a/sound/soc/pxa/em-x270.c +++ b/sound/soc/pxa/em-x270.c @@ -1,7 +1,7 @@ /* - * em-x270.c -- SoC audio for EM-X270 + * SoC audio driver for EM-X270, eXeda and CM-X300 * - * Copyright 2007 CompuLab, Ltd. + * Copyright 2007, 2009 CompuLab, Ltd. * * Author: Mike Rapoport <mike(a)compulab.co.il> * @@ -68,7 +68,8 @@ static int __init em_x270_init(void) { int ret; - if (!machine_is_em_x270()) + if (!(machine_is_em_x270() || machine_is_exeda() + || machine_is_cm_x300())) return -ENODEV; em_x270_snd_device = platform_device_alloc("soc-audio", -1); @@ -95,5 +96,5 @@ module_exit(em_x270_exit); /* Module information */ MODULE_AUTHOR("Mike Rapoport"); -MODULE_DESCRIPTION("ALSA SoC EM-X270"); +MODULE_DESCRIPTION("ALSA SoC EM-X270, eXeda and CM-X300"); MODULE_LICENSE("GPL"); -- 1.5.6.4

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[alsa-devel] [PATCH] ASoC: soc-core: fix crash when removing not instantiated card
by Mike Rapoport 11 May '09

11 May '09

If the card was not instantiated in snd_soc_instantiate_card, calling soc-remove will crash because some of codec, cpu_dai and card .remove methods are called twice. Fix this by returning from soc_remove immediately. Signed-off-by: Mike Rapoport <mike(a)compulab.co.il> --- sound/soc/soc-core.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 99712f6..1cd149b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -954,6 +954,9 @@ static int soc_remove(struct platform_device *pdev) struct snd_soc_platform *platform = card->platform; struct snd_soc_codec_device *codec_dev = socdev->codec_dev; + if (!card->instantiated) + return 0; + run_delayed_work(&card->delayed_work); if (platform->remove) -- 1.5.6.4

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[alsa-devel] Problem with sound on a Dell Dimension 4550 when it comes to using the ALSA-driver.
by Kristoffer Grundström 11 May '09

11 May '09

I was wondering if there will be a fix out for my problem that's described here: https://qa.mandriva.com/show_bug.cgi?id=49048 I'm getting sick and tired about having this problem in all the Mandriva-versions. Can you help me in any way? Do I have to add more info to that bugreport? /Kristoffer Grundstroem Unofficial Swedish translator of Mandriva.

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[alsa-devel] [PATCH] pxa2xx-ac97: fix reset gpio mode setting
by Mike Rapoport 11 May '09

11 May '09

Signed-off-by: Mike Rapoport <mike(a)compulab.co.il> --- sound/arm/pxa2xx-ac97-lib.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index a2c12d1..6fdca97 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -65,7 +65,7 @@ static void set_resetgpio_mode(int resetgpio_action) switch (resetgpio_action) { case RESETGPIO_NORMAL_ALTFUNC: if (reset_gpio == 113) - mode = 113 | GPIO_OUT | GPIO_DFLT_LOW; + mode = 113 | GPIO_ALT_FN_2_OUT; if (reset_gpio == 95) mode = 95 | GPIO_ALT_FN_1_OUT; break; -- 1.5.6.4

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[alsa-devel] [PATCH 0/1] ASoC: TWL4030: Possible fix for the Beep at the end of the audio file.
by Peter Ujfalusi 11 May '09

11 May '09

Hello, I have finally managed to reproduce and test the 'tuck' phenomena. The following patch seamingly fixes the issue at least in my hardware setup. Note that I still can not test this on Beagle and on the HW that I have testing the headset output is kind of tricky, so this might very well not fix the issue. Anuj, Jarkko, Arun: Would you be so kind and test it on your boards? --- Peter Ujfalusi (1): ASoC: TWL4030: Headset VMID ramp fix sound/soc/codecs/twl4030.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-)

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[alsa-devel] spdif/iec958 outputs at VT1708S not recognized
by Peter Niemayer 11 May '09

11 May '09

Hi, I recently installed an Asus M4N78 mainboard, that uses a nVidia GeForce 8300 chipset and a VT1708S soundchip. Alas, the code in pci/hda/patch_via.c did not recognize the 3 existing iec958/spdif outputs, and presented only the analog devices. I had to short circuit the pin definition in vt1708S_parse_auto_config() to get the digital outputs to work: spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + // test fails for Asus M4N78 if (spec->autocfg.dig_out_pin) spec->multiout.dig_out_nid = VT1708S_DIGOUT_NID; spec->extra_dig_out_nid = 0x15; The digital sound output works just fine for me with this change, but I guess there should be a cleaner way to make this work? Several people in forums on the InterNet reported similar problems with getting their digital outputs on VT1708S to work under Linux. Regards, Peter Niemayer

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[alsa-devel] speaker-test cleanups and fixes
by Dan McGee 11 May '09

11 May '09

This is a series of 3 (mostly unrelated) patches to clean up and fix up a few shortcomings I found in the speaker-test program. Please let me know of any concerns or comments on the patches. -Dan

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[alsa-devel] How to test new SDPIF implementation?
by Jon Smirl 10 May '09

10 May '09

I would like to test my SPDIF driver implementation. How can I used command line tools to play sample 5.1 music over SPDIF? I have a file 5.1 AC3 encoded file -- w.ac3 This doesn't work: iecset -D hw:0 audio off aplay -c 2 -r 48000 -f S16_BE -D hw:0,1 w.ac3 I can play uncompressed music on SDPIF without problem. I know my audio system will play the compressed music, I can use A52 from my desktop. -- Jon Smirl jonsmirl(a)gmail.com

2 1

[alsa-devel] [PATCH 1/2] speaker-test: move existing endian macros up in the file
by Dan McGee 10 May '09

10 May '09

This is necessary for a later patch removing the various endianness checks sprinkled throughout the code. Signed-off-by: Dan McGee <dpmcgee(a)gmail.com> --- speaker-test/speaker-test.c | 20 ++++++++++---------- 1 files changed, 10 insertions(+), 10 deletions(-) diff --git a/speaker-test/speaker-test.c b/speaker-test/speaker-test.c index d983a4f..c1d53b6 100644 --- a/speaker-test/speaker-test.c +++ b/speaker-test/speaker-test.c @@ -68,6 +68,16 @@ enum { #define MAX_CHANNELS 16 +#if __BYTE_ORDER == __LITTLE_ENDIAN +#define COMPOSE_ID(a,b,c,d) ((a) | ((b)<<8) | ((c)<<16) | ((d)<<24)) +#define LE_SHORT(v) (v) +#define LE_INT(v) (v) +#else +#define COMPOSE_ID(a,b,c,d) ((d) | ((c)<<8) | ((b)<<16) | ((a)<<24)) +#define LE_SHORT(v) bswap_16(v) +#define LE_INT(v) bswap_32(v) +#endif + static char *device = "default"; /* playback device */ static snd_pcm_format_t format = SND_PCM_FORMAT_S16; /* sample format */ static unsigned int rate = 48000; /* stream rate */ @@ -520,16 +530,6 @@ struct wave_header { } chunk; }; -#if __BYTE_ORDER == __LITTLE_ENDIAN -#define COMPOSE_ID(a,b,c,d) ((a) | ((b)<<8) | ((c)<<16) | ((d)<<24)) -#define LE_SHORT(v) (v) -#define LE_INT(v) (v) -#else -#define COMPOSE_ID(a,b,c,d) ((d) | ((c)<<8) | ((b)<<16) | ((a)<<24)) -#define LE_SHORT(v) bswap_16(v) -#define LE_INT(v) bswap_32(v) -#endif - #define WAV_RIFF COMPOSE_ID('R','I','F','F') #define WAV_WAVE COMPOSE_ID('W','A','V','E') #define WAV_FMT COMPOSE_ID('f','m','t',' ') -- 1.6.2.4

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Re: [alsa-devel] ASoC: pxa2xx-i2s
by Mark Brown 09 May '09

09 May '09

On Wed, Jan 21, 2009 at 10:46:17AM -0600, Brian G Rhodes wrote: > Mark Brown wrote: > >What is the clock master in your system? > The master is the i2s controller sysclk. The dac2x is in slave mode. > Though the i2s controller isn't directly supplying the bit clock to the > codec in this case. We're using the PLL on the dac2x. Just to confirm, the codec is the clock master on the I2S bus itself?

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[alsa-devel] [PATCH 3/4] pxa2xx-i2s: Cleaner use of the FIFOs
by Karl Beldan 09 May '09

09 May '09

The FIFO logic and the registers are reset at stream startup with SACR0_RST => the sibling function might suffer and the FIFOs might unpleasantly fill up whence the calls to pxa_i2s_wait all over the place. This gets rid of it. Signed-off-by: Karl Beldan <karl.beldan(a)mobile-devices.fr> --- sound/soc/pxa/pxa2xx-i2s.c | 20 +------------------- 1 files changed, 1 insertions(+), 19 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 0e53d07..755e2d1 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -106,25 +106,12 @@ static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream, if (IS_ERR(clk_i2s)) return PTR_ERR(clk_i2s); - if (!cpu_dai->active) { - SACR0 |= SACR0_RST; + if (!cpu_dai->active) SACR0 = 0; - } return 0; } -/* wait for I2S controller to be ready */ -static int pxa_i2s_wait(void) -{ - int i; - - /* flush the Rx FIFO */ - for(i = 0; i < 16; i++) - SADR; - return 0; -} - static int pxa2xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { @@ -169,7 +156,6 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, BUG_ON(IS_ERR(clk_i2s)); clk_enable(clk_i2s); - pxa_i2s_wait(); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) cpu_dai->dma_data = &pxa2xx_i2s_pcm_stereo_out; @@ -256,7 +242,6 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream, if ((SACR1 & (SACR1_DREC | SACR1_DRPL)) == (SACR1_DREC | SACR1_DRPL)) { SACR0 &= ~SACR0_ENB; - pxa_i2s_wait(); clk_disable(clk_i2s); } } @@ -275,7 +260,6 @@ static int pxa2xx_i2s_suspend(struct snd_soc_dai *dai) /* deactivate link */ SACR0 &= ~SACR0_ENB; - pxa_i2s_wait(); return 0; } @@ -284,8 +268,6 @@ static int pxa2xx_i2s_resume(struct snd_soc_dai *dai) if (!dai->active) return 0; - pxa_i2s_wait(); - SACR0 = pxa_i2s.sacr0 &= ~SACR0_ENB; SACR1 = pxa_i2s.sacr1; SAIMR = pxa_i2s.saimr; -- 1.6.3.rc1.34.g0be9b -- Karl

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[alsa-devel] [PATCH 1/4] pxa2xx-i2s: Proper initialization
by Karl Beldan 08 May '09

08 May '09

Reset FIFO logic and registers, and make sure REC and RPL functions and FIFO service are disabled. Signed-off-by: Karl Beldan <karl.beldan(a)mobile-devices.fr> --- sound/soc/pxa/pxa2xx-i2s.c | 12 ++++++++++++ 1 files changed, 12 insertions(+), 0 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 2f4b6e4..52862dc 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -370,6 +370,18 @@ static struct platform_driver pxa2xx_i2s_driver = { static int __init pxa2xx_i2s_init(void) { clk_i2s = ERR_PTR(-ENOENT); + + /* + * PXA Developer's Manual: + * If SACR0[ENB] is toggled in the middle of a normal operation, + * the SACR0[RST] bit must also be set and cleared to reset all + * I2S controller registers. + */ + SACR0 = SACR0_RST; + SACR0 = 0; + SACR1 = SACR1_DRPL | SACR1_DREC; + SAIMR &= ~(SAIMR_RFS | SAIMR_TFS); + return platform_driver_register(&pxa2xx_i2s_driver); } -- 1.6.3.rc1.34.g0be9b -- Karl

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[alsa-devel] [PATCH] AFEB9260 ASoC driver
by Sergey Lapin 08 May '09

08 May '09

ASoC driver for AT91SAM9260-based AFEB9260 board Signed-off-by: Sergey Lapin <slapin(a)ossfans.org> --- arch/arm/mach-at91/board-afeb-9260v1.c | 4 + sound/soc/atmel/Kconfig | 8 ++ sound/soc/atmel/Makefile | 1 + sound/soc/atmel/afeb9260.c | 217 ++++++++++++++++++++++++++++++++ 4 files changed, 230 insertions(+), 0 deletions(-) create mode 100644 sound/soc/atmel/afeb9260.c diff --git a/arch/arm/mach-at91/board-afeb-9260v1.c b/arch/arm/mach-at91/board-afeb-9260v1.c index e263fda..e3f1c56 100644 --- a/arch/arm/mach-at91/board-afeb-9260v1.c +++ b/arch/arm/mach-at91/board-afeb-9260v1.c @@ -164,6 +164,8 @@ static struct at91_mmc_data __initdata afeb9260_mmc_data = { static struct i2c_board_info __initdata afeb9260_i2c_devices[] = { { + I2C_BOARD_INFO("tlv320aic23", 0x1a), + }, { I2C_BOARD_INFO("fm3130", 0x68), }, { I2C_BOARD_INFO("24c64", 0x50), @@ -196,6 +198,8 @@ static void __init afeb9260_board_init(void) /* I2C */ at91_add_device_i2c(afeb9260_i2c_devices, ARRAY_SIZE(afeb9260_i2c_devices)); + /* Audio */ + at91_add_device_ssc(AT91SAM9260_ID_SSC, ATMEL_SSC_TX); } MACHINE_START(AFEB9260, "Custom afeb9260 board") diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index a608d70..e720d5e 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -41,3 +41,11 @@ config SND_AT32_SOC_PLAYPAQ_SLAVE and FRAME signals on the PlayPaq. Unless you want to play with the AT32 as the SSC master, you probably want to say N here, as this will give you better sound quality. + +config SND_AT91_SOC_AFEB9260 + tristate "SoC Audio support for AFEB9260 board" + depends on ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC + select SND_ATMEL_SOC_SSC + select SND_SOC_TLV320AIC23 + help + Say Y here to support sound on AFEB9260 board. diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile index f54a7cc..ac2df98 100644 --- a/sound/soc/atmel/Makefile +++ b/sound/soc/atmel/Makefile @@ -13,3 +13,4 @@ snd-soc-playpaq-objs := playpaq_wm8510.o obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o +obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += afeb9260.o diff --git a/sound/soc/atmel/afeb9260.c b/sound/soc/atmel/afeb9260.c new file mode 100644 index 0000000..1514d8d --- /dev/null +++ b/sound/soc/atmel/afeb9260.c @@ -0,0 +1,217 @@ +/* + * afeb9260.c -- SoC audio for AFEB9260 + * + * Copyright (C) 2008 Sergey Lapin <slapin(a)ossfans.org> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * 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. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/clk.h> +#include <linux/platform_device.h> + +#include <linux/atmel-ssc.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <linux/gpio.h> + +#include "../codecs/tlv320aic23.h" +#include "atmel-pcm.h" +#include "atmel_ssc_dai.h" + +#define CODEC_CLOCK 12000000 + +static int afeb9260_startup(struct snd_pcm_substream *substream) +{ + return 1; +} + +static void afeb9260_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + + dev_dbg(rtd->socdev->dev, "shutdown"); +} + +static int afeb9260_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 err; + + /* Set codec DAI configuration */ + err = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S| + SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBM_CFM); + if (err < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return err; + } + + /* Set cpu DAI configuration */ + err = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBM_CFM); + if (err < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return err; + } + + /* Set the codec system clock for DAC and ADC */ + err = + snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN); + + if (err < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return err; + } + + return err; +} + +static struct snd_soc_ops afeb9260_ops = { + .startup = afeb9260_startup, + .hw_params = afeb9260_hw_params, + .shutdown = afeb9260_shutdown, +}; + +static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"Headphone Jack", NULL, "LHPOUT"}, + {"Headphone Jack", NULL, "RHPOUT"}, + + {"LLINEIN", NULL, "Line In"}, + {"RLINEIN", NULL, "Line In"}, + + {"MICIN", NULL, "Mic Jack"}, +}; + +static int afeb9260_tlv320aic23_init(struct snd_soc_codec *codec) +{ + + /* Add osk5912 specific widgets */ + snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets, + ARRAY_SIZE(tlv320aic23_dapm_widgets)); + + /* Set up osk5912 specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Line In"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + + snd_soc_dapm_sync(codec); + + return 0; +} + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link afeb9260_dai = { + .name = "TLV320AIC23", + .stream_name = "AIC23", + .cpu_dai = &atmel_ssc_dai[0], + .codec_dai = &tlv320aic23_dai, + .init = afeb9260_tlv320aic23_init, + .ops = &afeb9260_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_machine_afeb9260 = { + .name = "AFEB9260", + .platform = &atmel_soc_platform, + .dai_link = &afeb9260_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device afeb9260_snd_devdata = { + .card = &snd_soc_machine_afeb9260, + .codec_dev = &soc_codec_dev_tlv320aic23, +}; + +static struct platform_device *afeb9260_snd_device; + +static int __init afeb9260_soc_init(void) +{ + int err; + struct device *dev; + struct atmel_ssc_info *ssc_p = afeb9260_dai.cpu_dai->private_data; + struct ssc_device *ssc = NULL; + + if (!(machine_is_afeb9260())) + return -ENODEV; + + ssc = ssc_request(0); + if (IS_ERR(ssc)) { + printk(KERN_ERR "ASoC: Failed to request SSC 0\n"); + err = PTR_ERR(ssc); + ssc = NULL; + goto err_ssc; + } + ssc_p->ssc = ssc; + + afeb9260_snd_device = platform_device_alloc("soc-audio", -1); + if (!afeb9260_snd_device) { + printk(KERN_ERR "ASoC: Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(afeb9260_snd_device, &afeb9260_snd_devdata); + afeb9260_snd_devdata.dev = &afeb9260_snd_device->dev; + err = platform_device_add(afeb9260_snd_device); + if (err) + goto err1; + + dev = &afeb9260_snd_device->dev; + + return 0; +err1: + platform_device_del(afeb9260_snd_device); + platform_device_put(afeb9260_snd_device); +err_ssc: + return err; + +} + +static void __exit afeb9260_soc_exit(void) +{ + platform_device_unregister(afeb9260_snd_device); +} + +module_init(afeb9260_soc_init); +module_exit(afeb9260_soc_exit); + +MODULE_AUTHOR("Sergey Lapin <slapin(a)ossfans.org>"); +MODULE_DESCRIPTION("ALSA SoC for AFEB9260"); +MODULE_LICENSE("GPL"); + -- 1.5.6.5

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[alsa-devel] ASoC updates for 2.6.31
by Mark Brown 08 May '09

08 May '09

The following changes since commit bbd993077d788589a86a718ba7a7895ba5e71a17: Mark Brown (1): ASoC: Remove redundant codec pointer from DAIs are available in the git repository at: ssh://master.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31 (git:// doesn't seem to be updating so...). Daniel Mack (2): ASoC: cs4270: introduce CS4270_I2C_INCR ASoC: cs4270: add power management support Mark Brown (2): ASoC: Add headers to match patterns in MAINTAINERS ASoC: Fix file patterns for PXA sound drivers Peter Ujfalusi (1): ASoC: TWL4030: Fix typo in twl4030_codec_mute function MAINTAINERS | 4 ++- sound/soc/codecs/cs4270.c | 61 ++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/twl4030.c | 2 +- 3 files changed, 63 insertions(+), 4 deletions(-)

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[alsa-devel] [PATCH 1/1] ASoC: TWL4030: Fix typo in twl4030_codec_mute function
by Peter Ujfalusi 08 May '09

08 May '09

Copy-paste error: TWL4030_PRECKL_GAIN >> TWL4030_PRECKR_GAIN It has not caused problems, since TWL4030_PRECKL_GAIN == TWL4030_PRECKR_GAIN == 0x30 Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/twl4030.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 1a00e4b..fd392c6 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -237,7 +237,7 @@ static void twl4030_codec_mute(struct snd_soc_codec *codec, int mute) TWL4030_REG_PRECKL_CTL); reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_PRECKR_CTL); twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, - reg_val & (~TWL4030_PRECKL_GAIN), + reg_val & (~TWL4030_PRECKR_GAIN), TWL4030_REG_PRECKR_CTL); /* Disable PLL */ -- 1.6.2.3

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[alsa-devel] [PATCH 1/1] ASoC: Added OMAP3 EVM support in ASoC.
by Anuj Aggarwal 08 May '09

08 May '09

Signed-off-by: Anuj Aggarwal <anuj.aggarwal(a)ti.com> --- sound/soc/omap/Kconfig | 8 +++ sound/soc/omap/Makefile | 2 + sound/soc/omap/omap3evm.c | 147 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 157 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/omap3evm.c diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 675732e..b771238 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -39,6 +39,14 @@ config SND_OMAP_SOC_OMAP2EVM help Say Y if you want to add support for SoC audio on the omap2evm board. +config SND_OMAP_SOC_OMAP3EVM + tristate "SoC Audio support for OMAP3EVM board" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3EVM + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on the omap3evm board. + config SND_OMAP_SOC_SDP3430 tristate "SoC Audio support for Texas Instruments SDP3430" depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_3430SDP diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 0c9e4ac..a37f498 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -10,6 +10,7 @@ snd-soc-n810-objs := n810.o snd-soc-osk5912-objs := osk5912.o snd-soc-overo-objs := overo.o snd-soc-omap2evm-objs := omap2evm.o +snd-soc-omap3evm-objs := omap3evm.o snd-soc-sdp3430-objs := sdp3430.o snd-soc-omap3pandora-objs := omap3pandora.o snd-soc-omap3beagle-objs := omap3beagle.o @@ -18,6 +19,7 @@ obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o +obj-$(CONFIG_MACH_OMAP3EVM) += snd-soc-omap3evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c new file mode 100644 index 0000000..2af5d93 --- /dev/null +++ b/sound/soc/omap/omap3evm.c @@ -0,0 +1,147 @@ +/* + * omap3evm.c -- ALSA SoC support for OMAP3 EVM + * + * Author: Anuj Aggarwal <anuj.aggarwal(a)ti.com> + * + * Based on sound/soc/omap/beagle.c by Steve Sakoman + * + * Copyright (C) 2008 Texas Instruments, Incorporated + * + * 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 version 2. + * + * This program is distributed "as is" WITHOUT ANY WARRANTY of any kind, + * whether express or implied; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <mach/gpio.h> +#include <mach/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/twl4030.h" + +static int omap3evm_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 ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "Can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "Can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "Can't set codec system clock\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops omap3evm_ops = { + .hw_params = omap3evm_hw_params, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link omap3evm_dai = { + .name = "TWL4030", + .stream_name = "TWL4030", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai, + .ops = &omap3evm_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_omap3evm = { + .name = "omap3evm", + .platform = &omap_soc_platform, + .dai_link = &omap3evm_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device omap3evm_snd_devdata = { + .card = &snd_soc_omap3evm, + .codec_dev = &soc_codec_dev_twl4030, +}; + +static struct platform_device *omap3evm_snd_device; + +static int __init omap3evm_soc_init(void) +{ + int ret; + + if (!machine_is_omap3evm()) { + pr_debug("Not OMAP3 EVM!\n"); + return -ENODEV; + } + pr_info("OMAP3 EVM SoC init\n"); + + omap3evm_snd_device = platform_device_alloc("soc-audio", -1); + if (!omap3evm_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(omap3evm_snd_device, &omap3evm_snd_devdata); + omap3evm_snd_devdata.dev = &omap3evm_snd_device->dev; + *(unsigned int *)omap3evm_dai.cpu_dai->private_data = 1; + + ret = platform_device_add(omap3evm_snd_device); + if (ret) + goto err1; + + return 0; + +err1: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(omap3evm_snd_device); + + return ret; +} + +static void __exit omap3evm_soc_exit(void) +{ + platform_device_unregister(omap3evm_snd_device); +} + +module_init(omap3evm_soc_init); +module_exit(omap3evm_soc_exit); + +MODULE_AUTHOR("Anuj Aggarwal <anuj.aggarwal(a)ti.com>"); +MODULE_DESCRIPTION("ALSA SoC OMAP3 EVM"); +MODULE_LICENSE("GPLv2"); -- 1.6.2.4

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[alsa-devel] asus xonar DX
by Dmitri Seletski 08 May '09

08 May '09

Hello boys and girls. I have successfully installed asus xonar DX onto PCIE x16 slot on Linux 2.6.28-gentoo-r5 SMP on amd64 architecture. Version of alsa package - alsa-driver1.0.20, alsa firmware 1.0.19. I used alsa package and not kernel modules. Several channels seem to work fine(my speakers are broken, no back channels available, started holy war with creative about it, guess with fiasco with faulty speakers and xfi drivers i am supporter of creative no more, i am not gonna bad mouth them either just yet). So far so good. Microphone works, surround sound works too, I am pretty happy. Thanks for all people who advised me something. If i come across this problem - you will hear from me again ;) Specially

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Re: [alsa-devel] 2.6.29 regression: left audio channel broken after resume from suspend with Intel HDA
by Takashi Iwai 08 May '09

08 May '09

At Tue, 21 Apr 2009 20:39:36 +0200, Tino Keitel wrote: > > On Tue, Apr 21, 2009 at 09:39:13 +0200, Takashi Iwai wrote: > > At Tue, 21 Apr 2009 09:32:19 +0200, > > Tino Keitel wrote: > > > > > > On Tue, Apr 21, 2009 at 07:24:40 +0200, Takashi Iwai wrote: > > > > The commit is fcad94a4c71c36a05f4d5c6dcb174534b4e0b136: > > > > ALSA: hda - Fix the cmd cache keys for amp verbs > > > > > > Thanks for the reply. However, the patch doesn't work. I just got > > > distorted sound again. > > > > Then please run alsa-info.sh (with --no-upload option) before and > > after suspend, and attach the generated files. Also, check the latest > > Linus tree whether the problem still appears. > > My first impression with suspend seems to be wrong. It also sometimes > happens just after a while, event without suspend. > > Attached are 2 alsa-info.sh outputs. One was taken after I noticed the > distorted sound, one after I reloaded snd-hda-intel to fix the > distortion. The mixer seems to be quite different after module reload. The codec communication looks a bit unstable on this device... Does this still happen on the latest 2.6.30 version? Takashi

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[alsa-devel] [PATCH 4/4] pxa2xx-i2s: Fix suspend/resume
by Karl Beldan 08 May '09

08 May '09

pxa2xx_i2s_resume is : - unconditionnaly setting SACR0_ENB - unsetting SACR0_ENB in pxa_i2s.sacr0 fix these. In pxa2xx_i2s_{resume,suspend}, save/restore registers even when !dai->active. Signed-off-by: Karl Beldan <karl.beldan(a)mobile-devices.fr> --- sound/soc/pxa/pxa2xx-i2s.c | 11 +++-------- 1 files changed, 3 insertions(+), 8 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 755e2d1..5b0f0bf 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -249,9 +249,6 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream, #ifdef CONFIG_PM static int pxa2xx_i2s_suspend(struct snd_soc_dai *dai) { - if (!dai->active) - return 0; - /* store registers */ pxa_i2s.sacr0 = SACR0; pxa_i2s.sacr1 = SACR1; @@ -265,14 +262,12 @@ static int pxa2xx_i2s_suspend(struct snd_soc_dai *dai) static int pxa2xx_i2s_resume(struct snd_soc_dai *dai) { - if (!dai->active) - return 0; - - SACR0 = pxa_i2s.sacr0 &= ~SACR0_ENB; + SACR0 = pxa_i2s.sacr0 & ~SACR0_ENB; SACR1 = pxa_i2s.sacr1; SAIMR = pxa_i2s.saimr; SADIV = pxa_i2s.sadiv; - SACR0 |= SACR0_ENB; + + SACR0 = pxa_i2s.sacr0; return 0; } -- 1.6.3.rc1.34.g0be9b -- Karl

1 0

[alsa-devel] [PATCH 0/4] pxa2xx-i2s: Rework
by Karl Beldan 08 May '09

08 May '09

Following is a rework of the i2s code for the pxa. I tested it with the ac97 clock as the master clock. Tests feedbacks are welcome. -- Karl

1 0

[alsa-devel] [PATCH 0/1] ASoC: Added OMAP3 EVM Support
by Anuj Aggarwal 07 May '09

07 May '09

With refernce to the discussion at: http://marc.info/?l=linux-omap&m=124169330310075&w=2 I am resending the patch to add OMAP3 EVM support in ASoC. It has been built/checked against the tip. Anuj Aggarwal (1): ASoC: Added OMAP3 EVM support in ASoC. sound/soc/omap/Kconfig | 8 +++ sound/soc/omap/Makefile | 2 + sound/soc/omap/omap3evm.c | 147 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 157 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/omap3evm.c

1 0

[alsa-devel] Laptop (Samsung Q45) speakers unmute when master channel is unmuted even with headphones connected
by Sam Morris 07 May '09

07 May '09

Since Linux 2.6.28 and through to 2.6.30-rc4, the hda-intel driver has had a bug on my laptop (a Samsung Q45). If I connect my headphones, the laptop's internal speakers are muted--fine. But if I then mute and unmute the master mixer channel, the speakers come back to life, even though the headphones are still connected. Please keep 517241-forwarded(a)bugs.debian.org CC'd in follow-ups. -- Sam Morris https://robots.org.uk/ PGP key id 1024D/5EA01078 3412 EA18 1277 354B 991B C869 B219 7FDB 5EA0 1078

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