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[alsa-devel] [PATCH] ASoC: count reaches 10001, not 10000.
by Roel Kluin 08 Feb '09

08 Feb '09

With a postfix increment count reaches 10001, not 10000. diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c index 7f2a5e1..edfbdc0 100644 --- a/sound/soc/blackfin/bf5xx-ad73311.c +++ b/sound/soc/blackfin/bf5xx-ad73311.c @@ -114,7 +114,7 @@ static int snd_ad73311_configure(void) SSYNC(); /* When TUVF is set, the data is already send out */ - while (!(status & TUVF) && count++ < 10000) { + while (!(status & TUVF) && ++count < 10000) { udelay(1); status = bfin_read_SPORT_STAT(); SSYNC(); @@ -123,7 +123,7 @@ static int snd_ad73311_configure(void) SSYNC(); local_irq_enable(); - if (count == 10000) { + if (count >= 10000) { printk(KERN_ERR "ad73311: failed to configure codec\n"); return -1; }

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[alsa-devel] [PATCH] Add initial support of Mitac mioa701 device SoC.
by Robert Jarzmik 08 Feb '09

08 Feb '09

This machine driver enables sound functions on Mitac mio a701 smartphone. Build upon ASoC v1, it handles : - rear speaker - front speaker - microphone - GSM A global "Mio Mode" switch is not yet provided to cope with audio path setup. As balance on audio chip line is no more assured, an incorrect setup can produce a lot of heat and even fry the battery behind the wm9713 and the speaker amplificator. It doesn't cope with : - headset jack - mio master mode - master volume control This driver is backported from ASoc v2, and amputated from scenario setups and master volume control. Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> --- sound/soc/pxa/Kconfig | 9 ++ sound/soc/pxa/Makefile | 2 + sound/soc/pxa/mioa701_wm9713.c | 250 ++++++++++++++++++++++++++++++++++++++++ 3 files changed, 261 insertions(+), 0 deletions(-) create mode 100644 sound/soc/pxa/mioa701_wm9713.c diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index f82e106..b385404 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -97,3 +97,12 @@ config SND_SOC_ZYLONITE help Say Y if you want to add support for SoC audio on the Marvell Zylonite reference platform. + +config SND_PXA2XX_SOC_MIOA701 + tristate "SoC Audio support for MIO A701" + depends on SND_PXA2XX_SOC && MACH_MIOA701 + select SND_PXA2XX_SOC_AC97 + select SND_SOC_WM9713 + help + Say Y if you want to add support for SoC audio on the + MIO A701. diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 08a9f27..9a6d27c 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -18,6 +18,7 @@ snd-soc-spitz-objs := spitz.o snd-soc-em-x270-objs := em-x270.o snd-soc-palm27x-objs := palm27x.o snd-soc-zylonite-objs := zylonite.o +snd-soc-mioa701-objs := mioa701_wm9713.o obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o @@ -26,4 +27,5 @@ obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o +obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c new file mode 100644 index 0000000..839f928 --- /dev/null +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -0,0 +1,250 @@ +/* + * Handles the Mitac mioa701 SoC system + * + * Copyright (C) 2008 Robert Jarzmik + * + * 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 in version 2 of the License. + * + * 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + * This is a little schema of the sound interconnections : + * + * Sagem X200 Wolfson WM9713 + * +--------+ +-------------------+ Rear Speaker + * | | | | /-+ + * | +--->----->---+MONOIN SPKL+--->----+-+ | + * | GSM | | | | | | + * | +--->----->---+PCBEEP SPKR+--->----+-+ | + * | CHIP | | | \-+ + * | +---<-----<---+MONO | + * | | | | Front Speaker + * +--------+ | | /-+ + * | HPL+--->----+-+ | + * | | | | | + * | OUT3+--->----+-+ | + * | | \-+ + * | | + * | | Front Micro + * | | + + * | MIC1+-----<--+o+ + * | | + + * +-------------------+ --- + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/platform_device.h> + +#include <asm/mach-types.h> +#include <mach/audio.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/ac97_codec.h> + +#include "pxa2xx-pcm.h" +#include "pxa2xx-ac97.h" +#include "../codecs/wm9713.h" + +#define ARRAY_AND_SIZE(x) (x), ARRAY_SIZE(x) + +#define AC97_GPIO_PULL 0x58 + +/* Use GPIO8 for rear speaker amplificator */ +static int rear_amp_power(struct snd_soc_codec *codec, int power) +{ + unsigned short reg; + + if (power) { + reg = snd_soc_read(codec, AC97_GPIO_CFG); + snd_soc_write(codec, AC97_GPIO_CFG, reg | 0x0100); + reg = snd_soc_read(codec, AC97_GPIO_PULL); + snd_soc_write(codec, AC97_GPIO_PULL, reg | (1<<15)); + } else { + reg = snd_soc_read(codec, AC97_GPIO_CFG); + snd_soc_write(codec, AC97_GPIO_CFG, reg & ~0x0100); + reg = snd_soc_read(codec, AC97_GPIO_PULL); + snd_soc_write(codec, AC97_GPIO_PULL, reg & ~(1<<15)); + } + + return 0; +} + +static int rear_amp_event(struct snd_soc_dapm_widget *widget, + struct snd_kcontrol *kctl, int event) +{ + struct snd_soc_codec *codec = widget->codec; + + return rear_amp_power(codec, SND_SOC_DAPM_EVENT_ON(event)); +} + +/* mioa701 machine dapm widgets */ +static const struct snd_soc_dapm_widget mioa701_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Front Speaker", NULL), + SND_SOC_DAPM_SPK("Rear Speaker", rear_amp_event), + SND_SOC_DAPM_MIC("Headset", NULL), + SND_SOC_DAPM_LINE("GSM Line Out", NULL), + SND_SOC_DAPM_LINE("GSM Line In", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Front Mic", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Call Mic */ + {"Mic Bias", NULL, "Front Mic"}, + {"MIC1", NULL, "Mic Bias"}, + + /* Headset Mic */ + {"LINEL", NULL, "Headset Mic"}, + {"LINER", NULL, "Headset Mic"}, + + /* GSM Module */ + {"MONOIN", NULL, "GSM Line Out"}, + {"PCBEEP", NULL, "GSM Line Out"}, + {"GSM Line In", NULL, "MONO"}, + + /* headphone connected to HPL, HPR */ + {"Headset", NULL, "HPL"}, + {"Headset", NULL, "HPR"}, + + /* front speaker connected to HPL, OUT3 */ + {"Front Speaker", NULL, "HPL"}, + {"Front Speaker", NULL, "OUT3"}, + + /* rear speaker connected to SPKL, SPKR */ + {"Rear Speaker", NULL, "SPKL"}, + {"Rear Speaker", NULL, "SPKR"}, +}; + +static int mioa701_wm9713_init(struct snd_soc_codec *codec) +{ + unsigned short reg; + + /* Add mioa701 specific widgets */ + snd_soc_dapm_new_controls(codec, ARRAY_AND_SIZE(mioa701_dapm_widgets)); + + /* Set up mioa701 specific audio path audio_mapnects */ + snd_soc_dapm_add_routes(codec, ARRAY_AND_SIZE(audio_map)); + + /* Prepare GPIO8 for rear speaker amplificator */ + reg = codec->read(codec, AC97_GPIO_CFG); + codec->write(codec, AC97_GPIO_CFG, reg | 0x0100); + + /* Prepare MIC input */ + reg = codec->read(codec, AC97_3D_CONTROL); + codec->write(codec, AC97_3D_CONTROL, reg | 0xc000); + + snd_soc_dapm_enable_pin(codec, "Front Speaker"); + snd_soc_dapm_enable_pin(codec, "Rear Speaker"); + snd_soc_dapm_enable_pin(codec, "Front Mic"); + snd_soc_dapm_enable_pin(codec, "GSM Line In"); + snd_soc_dapm_enable_pin(codec, "GSM Line Out"); + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_ops mioa701_ops; + +static struct snd_soc_dai_link mioa701_dai[] = { + { + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI], + .codec_dai = &wm9713_dai[WM9713_DAI_AC97_HIFI], + .init = mioa701_wm9713_init, + .ops = &mioa701_ops, + }, + { + .name = "AC97 Aux", + .stream_name = "AC97 Aux", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX], + .codec_dai = &wm9713_dai[WM9713_DAI_AC97_AUX], + .ops = &mioa701_ops, + }, +}; + +static struct snd_soc_card mioa701 = { + .name = "MioA701", + .platform = &pxa2xx_soc_platform, + .dai_link = mioa701_dai, + .num_links = ARRAY_SIZE(mioa701_dai), +}; + +static struct snd_soc_device mioa701_snd_devdata = { + .card = &mioa701, + .codec_dev = &soc_codec_dev_wm9713, +}; + +static struct platform_device *mioa701_snd_device; + +static int mioa701_wm9713_probe(struct platform_device *pdev) +{ + int ret; + + if (!machine_is_mioa701()) + return -ENODEV; + + dev_warn(&pdev->dev, "Be warned that incorrect mixes/muxes setup will" + "lead to overheating and possible destruction of your device." + "Do not use without a good knowledge of mio's board design!\n"); + + mioa701_snd_device = platform_device_alloc("soc-audio", -1); + if (!mioa701_snd_device) + return -ENOMEM; + + platform_set_drvdata(mioa701_snd_device, &mioa701_snd_devdata); + mioa701_snd_devdata.dev = &mioa701_snd_device->dev; + + ret = platform_device_add(mioa701_snd_device); + if (!ret) + return 0; + + platform_device_put(mioa701_snd_device); + return ret; +} + +static int __devexit mioa701_wm9713_remove(struct platform_device *pdev) +{ + platform_device_unregister(mioa701_snd_device); + return 0; +} + +static struct platform_driver mioa701_wm9713_driver = { + .probe = mioa701_wm9713_probe, + .remove = __devexit_p(mioa701_wm9713_remove), + .driver = { + .name = "mioa701-wm9713", + .owner = THIS_MODULE, + }, +}; + +static int __init mioa701_asoc_init(void) +{ + return platform_driver_register(&mioa701_wm9713_driver); +} + +static void __exit mioa701_asoc_exit(void) +{ + platform_driver_unregister(&mioa701_wm9713_driver); +} + +module_init(mioa701_asoc_init); +module_exit(mioa701_asoc_exit); + +/* Module information */ +MODULE_AUTHOR("Robert Jarzmik (rjarzmik(a)free.fr)"); +MODULE_DESCRIPTION("ALSA SoC WM9713 MIO A701"); +MODULE_LICENSE("GPL"); -- 1.5.6.5

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[alsa-devel] Thank for your efforts.
by Guy Stalnaker 08 Feb '09

08 Feb '09

Takashi, Herton, Tim, Jeanne-Pierre, etc. My thanks for your efforts to get sound working on the internal speakers on HP Pavilion Dv4 laptops. I have a brand new1225 (one week old) and Ubuntu 8.10 works splendidly in all the things I need (wireless, graphics, etc.). Sound works, however, only on the front headphone jack, but not through the internal speakers. I offer myself as another test platform for your test code. To date I've installed the lastest snapshot, have model=hp-dv5 in /etc/modprobe.d/alsa-base, and I've tried Takashi's hda-verb option to modify the pin. That does change the pin value, but it has no affect, internal speaker's remain dead. Best regards, Guy -- Guy Stalnaker, I^2@DOIT, 1210 West Dayton Street, Room 3209 CSS, Madison WI 53719-1220, jstalnak(a)wisc.edu, work 608.263.8035, cell 608.235.4718, fax 608.265.6681, page page-guy(a)watchdog.doit.wisc.edu

1 0

[alsa-devel] RME HDSP rev 37 does not work on 64-bit linux
by Dmitry Baikov 07 Feb '09

07 Feb '09

Hello, recently I upgraded firmware for my RME HDSP (Cardbus + Multiface II) to recent version, to be able to use it under WinXP 64, since there's no 64-bit windows driver for old hdsp firmware. And the results under linux are surprising: My main OS is Ubuntu 8.10 64-bit, and there hdspmixer fails with the message: Card 1 : RME Hammerfall DSP at 0x130000000, irq 18 Uninitialized HDSP card found. Use hdsploader to upload configuration data to the card. No Hammerfall DSP card found. Trying to run hdsploader gives: Card 1 : RME Hammerfall DSP at 0x130000000, irq 18 Upload firmware for card hw:1 Hwdep ioctl error on card hw:1 : Input/output error. dmesg: Hammerfall-DSP: wait for FIFO status <= 0 failed after 30 iterations This happens even if I boot into winxp first to initialize multiface, then reboot into linux and do not run hdsploader. BUT: under 32-bit ubuntu 8.10 with the same kernel 2.6.27-11-generic and the same version of alsa-*: 1.0.17-0ubuntu1, everything works fine both with cold boot + hdsploader and winxp -> reboot-> no hdsploader. Regards, Dmitry.

4 8

[alsa-devel] redirect stream of audio
by Hartman Mario Andres 07 Feb '09

07 Feb '09

1 0

[PATCH] hda: Add mising beep controls for realtek alc268 codec.
by Kusanagi Kouichi 07 Feb '09

07 Feb '09

So far, some models didn't add a beep control since they don't use analog beep in. Signed-off-by: Kusanagi Kouichi <slash(a)ma.neweb.ne.jp> --- sound/pci/hda/patch_realtek.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f594a09..2fa26f9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11887,7 +11887,7 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = { static struct alc_config_preset alc268_presets[] = { [ALC267_QUANTA_IL1] = { - .mixers = { alc267_quanta_il1_mixer }, + .mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer }, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc267_quanta_il1_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), @@ -11969,7 +11969,7 @@ static struct alc_config_preset alc268_presets[] = { }, [ALC268_ACER_ASPIRE_ONE] = { .mixers = { alc268_acer_aspire_one_mixer, - alc268_capture_alt_mixer }, + alc268_capture_alt_mixer, alc268_beep_mixer }, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc268_acer_aspire_one_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), -- 1.5.6.5 --UlVJffcvxoiEqYs2--

1 0

[alsa-devel] Internal Mic of Philips SPC 1300NC Webcam not recording (same with SPC 1000 and 620)
by Sinisa Dukanovic 07 Feb '09

07 Feb '09

Hi everyone, i can't record sound from the internal mic of my Phillips SPC1300 Webcam. I have already filed a bug report at: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3927 where i posted a lot of information. In addition to that, you can find the latest results of alsa-info.sh here: http://www.alsa-project.org/db/?f=a3fabdb13269a50e5068a76b713a0d45f1794aaf Is there anything (besides those infos in the bugreport) that i can do to help fixing this bug? regards, SD

1 0

[alsa-devel] [PATCH 2.6.30] ASoC: optimize init sequence of Freescale MPC8610 sound drivers
by Timur Tabi 06 Feb '09

06 Feb '09

In the Freescale MPC8610 sound drivers, relocate all code from the _prepare functions into the corresponding _hw_params functions. These drivers assumed that the sample size is known in the _prepare function and not in the _hw_params function, but this is not true. Move the code in fsl_dma_prepare() into fsl_dma_hw_param(). Create fsl_ssi_hw_params() and move the code from fsl_ssi_prepare() into it. Turn off snooping for DMA operations to/from I/O registers, since that's not necessary. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/fsl_dma.c | 178 ++++++++++++++++++++++------------------------ sound/soc/fsl/fsl_ssi.c | 19 +++--- 2 files changed, 94 insertions(+), 103 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 64993ed..58a3fa4 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -464,11 +464,7 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) sizeof(struct fsl_dma_link_descriptor); for (i = 0; i < NUM_DMA_LINKS; i++) { - struct fsl_dma_link_descriptor *link = &dma_private->link[i]; - - link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); - link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); - link->next = cpu_to_be64(temp_link); + dma_private->link[i].next = cpu_to_be64(temp_link); temp_link += sizeof(struct fsl_dma_link_descriptor); } @@ -525,79 +521,9 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) * This function obtains hardware parameters about the opened stream and * programs the DMA controller accordingly. * - * Note that due to a quirk of the SSI's STX register, the target address - * for the DMA operations depends on the sample size. So we don't program - * the dest_addr (for playback -- source_addr for capture) fields in the - * link descriptors here. We do that in fsl_dma_prepare() - */ -static int fsl_dma_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct fsl_dma_private *dma_private = runtime->private_data; - - dma_addr_t temp_addr; /* Pointer to next period */ - - unsigned int i; - - /* Get all the parameters we need */ - size_t buffer_size = params_buffer_bytes(hw_params); - size_t period_size = params_period_bytes(hw_params); - - /* Initialize our DMA tracking variables */ - dma_private->period_size = period_size; - dma_private->num_periods = params_periods(hw_params); - dma_private->dma_buf_end = dma_private->dma_buf_phys + buffer_size; - dma_private->dma_buf_next = dma_private->dma_buf_phys + - (NUM_DMA_LINKS * period_size); - if (dma_private->dma_buf_next >= dma_private->dma_buf_end) - dma_private->dma_buf_next = dma_private->dma_buf_phys; - - /* - * The actual address in STX0 (destination for playback, source for - * capture) is based on the sample size, but we don't know the sample - * size in this function, so we'll have to adjust that later. See - * comments in fsl_dma_prepare(). - * - * The DMA controller does not have a cache, so the CPU does not - * need to tell it to flush its cache. However, the DMA - * controller does need to tell the CPU to flush its cache. - * That's what the SNOOP bit does. - * - * Also, even though the DMA controller supports 36-bit addressing, for - * simplicity we currently support only 32-bit addresses for the audio - * buffer itself. - */ - temp_addr = substream->dma_buffer.addr; - - for (i = 0; i < NUM_DMA_LINKS; i++) { - struct fsl_dma_link_descriptor *link = &dma_private->link[i]; - - link->count = cpu_to_be32(period_size); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - link->source_addr = cpu_to_be32(temp_addr); - else - link->dest_addr = cpu_to_be32(temp_addr); - - temp_addr += period_size; - } - - return 0; -} - -/** - * fsl_dma_prepare - prepare the DMA registers for playback. - * - * This function is called after the specifics of the audio data are known, - * i.e. snd_pcm_runtime is initialized. - * - * In this function, we finish programming the registers of the DMA - * controller that are dependent on the sample size. - * - * One of the drawbacks with big-endian is that when copying integers of - * different sizes to a fixed-sized register, the address to which the - * integer must be copied is dependent on the size of the integer. + * One drawback of big-endian is that when copying integers of different + * sizes to a fixed-sized register, the address to which the integer must be + * copied is dependent on the size of the integer. * * For example, if P is the address of a 32-bit register, and X is a 32-bit * integer, then X should be copied to address P. However, if X is a 16-bit @@ -613,22 +539,58 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream, * and 8 bytes at a time). So we do not support packed 24-bit samples. * 24-bit data must be padded to 32 bits. */ -static int fsl_dma_prepare(struct snd_pcm_substream *substream) +static int fsl_dma_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) { struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_dma_private *dma_private = runtime->private_data; + + /* Number of bits per sample */ + unsigned int sample_size = + snd_pcm_format_physical_width(params_format(hw_params)); + + /* Number of bytes per frame */ + unsigned int frame_size = 2 * (sample_size / 8); + + /* Bus address of SSI STX register */ + dma_addr_t ssi_sxx_phys = dma_private->ssi_sxx_phys; + + /* Size of the DMA buffer, in bytes */ + size_t buffer_size = params_buffer_bytes(hw_params); + + /* Number of bytes per period */ + size_t period_size = params_period_bytes(hw_params); + + /* Pointer to next period */ + dma_addr_t temp_addr = substream->dma_buffer.addr; + + /* Pointer to DMA controller */ struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel; - u32 mr; + + u32 mr; /* DMA Mode Register */ + unsigned int i; - dma_addr_t ssi_sxx_phys; /* Bus address of SSI STX register */ - unsigned int frame_size; /* Number of bytes per frame */ - ssi_sxx_phys = dma_private->ssi_sxx_phys; + /* Initialize our DMA tracking variables */ + dma_private->period_size = period_size; + dma_private->num_periods = params_periods(hw_params); + dma_private->dma_buf_end = dma_private->dma_buf_phys + buffer_size; + dma_private->dma_buf_next = dma_private->dma_buf_phys + + (NUM_DMA_LINKS * period_size); + + if (dma_private->dma_buf_next >= dma_private->dma_buf_end) + /* This happens if the number of periods == NUM_DMA_LINKS */ + dma_private->dma_buf_next = dma_private->dma_buf_phys; mr = in_be32(&dma_channel->mr) & ~(CCSR_DMA_MR_BWC_MASK | CCSR_DMA_MR_SAHTS_MASK | CCSR_DMA_MR_DAHTS_MASK); - switch (runtime->sample_bits) { + /* Due to a quirk of the SSI's STX register, the target address + * for the DMA operations depends on the sample size. So we calculate + * that offset here. While we're at it, also tell the DMA controller + * how much data to transfer per sample. + */ + switch (sample_size) { case 8: mr |= CCSR_DMA_MR_DAHTS_1 | CCSR_DMA_MR_SAHTS_1; ssi_sxx_phys += 3; @@ -641,12 +603,12 @@ static int fsl_dma_prepare(struct snd_pcm_substream *substream) mr |= CCSR_DMA_MR_DAHTS_4 | CCSR_DMA_MR_SAHTS_4; break; default: + /* We should never get here */ dev_err(substream->pcm->card->dev, - "unsupported sample size %u\n", runtime->sample_bits); + "unsupported sample size %u\n", sample_size); return -EINVAL; } - frame_size = runtime->frame_bits / 8; /* * BWC should always be a multiple of the frame size. BWC determines * how many bytes are sent/received before the DMA controller checks the @@ -655,7 +617,6 @@ static int fsl_dma_prepare(struct snd_pcm_substream *substream) * capture, the receive FIFO is triggered when it contains one frame, so * we want to receive one frame at a time. */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) mr |= CCSR_DMA_MR_BWC(2 * frame_size); else @@ -663,16 +624,48 @@ static int fsl_dma_prepare(struct snd_pcm_substream *substream) out_be32(&dma_channel->mr, mr); - /* - * Program the address of the DMA transfer to/from the SSI. - */ for (i = 0; i < NUM_DMA_LINKS; i++) { struct fsl_dma_link_descriptor *link = &dma_private->link[i]; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + link->count = cpu_to_be32(period_size); + + /* Even though the DMA controller supports 36-bit addressing, + * for simplicity we allow only 32-bit addresses for the audio + * buffer itself. This was enforced in fsl_dma_new() with the + * DMA mask. + * + * The snoop bit tells the DMA controller whether it should tell + * the ECM to snoop during a read or write to an address. For + * audio, we use DMA to transfer data between memory and an I/O + * device (the SSI's STX0 or SRX0 register). Snooping is only + * needed if there is a cache, so we need to snoop memory + * addresses only. For playback, that means we snoop the source + * but not the destination. For capture, we snoop the + * destination but not the source. + * + * Note that failing to snoop properly is unlikely to cause + * cache incoherency if the period size is larger than the + * size of L1 cache. This is because filling in one period will + * flush out the data for the previous period. So if you + * increased period_bytes_min to a large enough size, you might + * get more performance by not snooping, and you'll still be + * okay. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + link->source_addr = cpu_to_be32(temp_addr); + link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); + link->dest_addr = cpu_to_be32(ssi_sxx_phys); - else + link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_NOSNOOP); + } else { link->source_addr = cpu_to_be32(ssi_sxx_phys); + link->source_attr = cpu_to_be32(CCSR_DMA_ATR_NOSNOOP); + + link->dest_addr = cpu_to_be32(temp_addr); + link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); + } + + temp_addr += period_size; } return 0; @@ -808,7 +801,6 @@ static struct snd_pcm_ops fsl_dma_ops = { .ioctl = snd_pcm_lib_ioctl, .hw_params = fsl_dma_hw_params, .hw_free = fsl_dma_hw_free, - .prepare = fsl_dma_prepare, .pointer = fsl_dma_pointer, }; diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c6d6eb7..6844009 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -400,7 +400,7 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, } /** - * fsl_ssi_prepare: prepare the SSI. + * fsl_ssi_hw_params - program the sample size * * Most of the SSI registers have been programmed in the startup function, * but the word length must be programmed here. Unfortunately, programming @@ -412,20 +412,19 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, * Note: The SxCCR.DC and SxCCR.PM bits are only used if the SSI is the * clock master. */ -static int fsl_ssi_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) +static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params, struct snd_soc_dai *cpu_dai) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data; - - struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + struct fsl_ssi_private *ssi_private = cpu_dai->private_data; if (substream == ssi_private->first_stream) { + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + unsigned int sample_size = + snd_pcm_format_width(params_format(hw_params)); u32 wl; /* The SSI should always be disabled at this points (SSIEN=0) */ - wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format)); + wl = CCSR_SSI_SxCCR_WL(sample_size); /* In synchronous mode, the SSI uses STCCR for capture */ clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl); @@ -579,7 +578,7 @@ static struct snd_soc_dai fsl_ssi_dai_template = { }, .ops = { .startup = fsl_ssi_startup, - .prepare = fsl_ssi_prepare, + .hw_params = fsl_ssi_hw_params, .shutdown = fsl_ssi_shutdown, .trigger = fsl_ssi_trigger, .set_sysclk = fsl_ssi_set_sysclk, -- 1.5.5

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[alsa-devel] [2.6.29-rc3][PATCH 1/2] ASoC: TLV320AIC3X: Fix kcontrol's private value use in put callback
by Jarkko Nikula 06 Feb '09

06 Feb '09

From: Eero Nurkkala <ext-eero.nurkkala(a)nokia.com> Function snd_soc_dapm_put_volsw_aic3x misuses the kcontrol's private value by still accessing it as bitfields even SOC_SINGLE_VALUE constructs it as a pointer into struct soc_mixer_control after the commit 4eaa9819dc08d7bfd1065ce530e31b18a119dcaf. This was causing arbitrary register writes when touching the controls defined with SOC_DAPM_SINGLE_AIC3X. Signed-off-by: Eero Nurkkala <ext-eero.nurkkala(a)nokia.com> Acked-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> --- sound/soc/codecs/tlv320aic3x.c | 11 +++++++---- 1 files changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index b47a749..aea0cb7 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -165,10 +165,13 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value & 0xff; - int shift = (kcontrol->private_value >> 8) & 0x0f; - int mask = (kcontrol->private_value >> 16) & 0xff; - int invert = (kcontrol->private_value >> 24) & 0x01; + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + int max = mc->max; + unsigned int mask = (1 << fls(max)) - 1; + unsigned int invert = mc->invert; unsigned short val, val_mask; int ret; struct snd_soc_dapm_path *path; -- 1.5.6.5

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Re: [alsa-devel] S/PDIF-Out on Asus M3A-H/HDMI with ALC1200]
by Thomas Schneider 06 Feb '09

06 Feb '09

Hi, here is the output from alsa-info.sh http://www.alsa-project.org/db/?f=8282910dc807e1f141896886cd5f8fd574117000 I found another small mistake. In the Mixer my Soundcard is shown as "Realtek ALC888". When i switch this to lines in patch_realtek.c then it is shown as "Realtek ALC1200". static struct hda_codec_preset snd_hda_preset_realtek[] = { ...... { .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc883 }, { .id = 0x10ec0888, .rev = 0x100101, .name = "ALC1200", .patch = patch_alc883 }, ..... Thomas Am 25.12.2008 05:31, schrieb Wu Fengguang: > Hi Thomas, > > Could you provide more information by running > http://www.alsa-project.org/alsa-info.sh > > I'm particularity interested in the full dmesg and contents of > /proc/asound/card?/*\#? > > Thanks, > Fengguang > > On Tue, Dec 23, 2008 at 12:22:29PM +0100, Thomas Schneider wrote: > >> Hi, >> >> i am looking for a way to make Digital-Audio work with the external >> and internal S/PDIF-Output of my Motherboard Asus M3A-H/HDMI with >> ALC1200 Soundard. I use the latest unstable snapshot and i can´t >> make it work at the same time. >> >> On the external S/PDIF-Output is my Digital Receiver connected, >> on the internal one my GeForce 8500GT to get sound over HDMI >> on my LCT-TV. >> >> When i load the driver without any model-options my Soundcard is >> detected as Asus P5Q-EM HDMI >> >> ALSA /usr/local/src/alsa-driver-unstable/pci/hda/hda_codec.c:2716: >> hda_codec: model 'asus-p5q' is selected for config 1043:82fe (Asus >> P5Q-EM HDMI) >> [ 6.721700] ALSA >> /usr/local/src/alsa-driver-unstable/pci/hda/hda_codec.c:1339: Cannot >> find slave Headphone Playback Volume, skipped >> [ 6.721714] ALSA >> /usr/local/src/alsa-driver-unstable/pci/hda/hda_codec.c:1339: Cannot >> find slave Speaker Playback Volume, skipped >> [ 6.721716] ALSA >> /usr/local/src/alsa-driver-unstable/pci/hda/hda_codec.c:1339: Cannot >> find slave Mono Playback Volume, skipped >> [ 6.721719] ALSA >> /usr/local/src/alsa-driver-unstable/pci/hda/hda_codec.c:1339: Cannot >> find slave Line-Out Playback Volume, skipped >> [ 6.721721] ALSA >> /usr/local/src/alsa-driver-unstable/pci/hda/hda_codec.c:1339: Cannot >> find slave PCM Playback Volume, skipped >> [ 6.721729] ALSA >> /usr/local/src/alsa-driver-unstable/pci/hda/hda_codec.c:1339: Cannot >> find slave Speaker Playback Switch, skipped >> [ 6.721731] ALSA >> /usr/local/src/alsa-driver-unstable/pci/hda/hda_codec.c:1339: Cannot >> find slave Mono Playback Switch, skipped >> >> Now i have sound on my LCD-TV over HDMI (internal S/PDIF-Out) >> >> When i load the driver with model=6stack-dig i have sound on my >> Digital-Receiver (external S/PDIF-Out). >> >> In the mailinglist-archive or in the doku i have seen, that the only >> difference >> between this models is the DIGOUT_NID >> >> asus-p5q = 0x10 >> 6stack-dig = 0x06 >> >> I there a way to get both output to work at the same time so i have >> sound on my LCD-TV and my Digital-Receiver ? >> >> >> Regards >> >> Thomas >> _______________________________________________ >> Alsa-devel mailing list >> Alsa-devel(a)alsa-project.org >> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >> > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >

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