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

[alsa-devel] ALSA on Alienware M17x
by Adam Gray 06 Dec '09

06 Dec '09

Hi guys, I'm having a lot of trouble getting sound out of my Alienware M17x. I've attached the output from alsa-info. I'm running kernel version 2.6.28 with ALSA driver version 1.0.18. There's no sound at all from the speakers or the headphones. In alsamixer, all channels appear un-muted and turned up full. If you need any further information please let me know. Regards, Adam Gray

3 19

[alsa-devel] [PATCH 2/4] ASoC: DaVinci: Voice Codec Interface
by miguel.aguilar＠ridgerun.com 25 Nov '09

25 Nov '09

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> 1) Adds an interface needed by the voice codec CQ0093. 2) Add an option to select internal or external codec in the DM365. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- sound/soc/davinci/Kconfig | 26 +++- sound/soc/davinci/Makefile | 2 + sound/soc/davinci/davinci-vcif.c | 363 ++++++++++++++++++++++++++++++++++++++ sound/soc/davinci/davinci-vcif.h | 63 +++++++ 4 files changed, 452 insertions(+), 2 deletions(-) create mode 100644 sound/soc/davinci/davinci-vcif.c create mode 100644 sound/soc/davinci/davinci-vcif.h diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 047ee39..051a10e 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -9,18 +9,40 @@ config SND_DAVINCI_SOC config SND_DAVINCI_SOC_I2S tristate +config SND_DAVINCI_SOC_VCIF + tristate + config SND_DAVINCI_SOC_MCASP tristate config SND_DAVINCI_SOC_EVM tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM" depends on SND_DAVINCI_SOC - depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM + depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM select SND_DAVINCI_SOC_I2S select SND_SOC_TLV320AIC3X help Say Y if you want to add support for SoC audio on TI - DaVinci DM6446 or DM355 EVM platforms. + DaVinci DM6446, DM355 or DM365 EVM platforms. + +choice + prompt "DM365 codec select" + depends on SND_DAVINCI_SOC_EVM + depends on MACH_DAVINCI_DM365_EVM + default SND_DM365_EXTERNAL_CODEC + +config SND_DM365_INTERNAL_CODEC + bool "CQ93VC" + select SND_DAVINCI_SOC_VCIF + select SND_SOC_CQ0093VC + help + Say Y if you want to add support for SoC On-chip voice codec + +config SND_DM365_EXTERNAL_CODEC + bool "AIC3101" + help + Say Y if you want to add support for AIC3101 audio codec +endchoice config SND_DM6467_SOC_EVM tristate "SoC Audio support for DaVinci DM6467 EVM" diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index a6939d7..a93679d 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -2,10 +2,12 @@ snd-soc-davinci-objs := davinci-pcm.o snd-soc-davinci-i2s-objs := davinci-i2s.o snd-soc-davinci-mcasp-objs:= davinci-mcasp.o +snd-soc-davinci-vcif-objs:= davinci-vcif.o obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o obj-$(CONFIG_SND_DAVINCI_SOC_MCASP) += snd-soc-davinci-mcasp.o +obj-$(CONFIG_SND_DAVINCI_SOC_VCIF) += snd-soc-davinci-vcif.o # DAVINCI Machine Support snd-soc-evm-objs := davinci-evm.o diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c new file mode 100644 index 0000000..83191f1 --- /dev/null +++ b/sound/soc/davinci/davinci-vcif.c @@ -0,0 +1,363 @@ +/* + * ALSA SoC Voice Codec Interface for TI DAVINCI processor + * + * Copyright (C) 2009 Texas Instruments. + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * Initial code: Hui Geng + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * 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 + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/clk.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "davinci-pcm.h" +#include "davinci-i2s.h" +#include "davinci-vcif.h" + +#define MOD_REG_BIT(val, mask, set) do { \ + if (set) { \ + val |= mask; \ + } else { \ + val &= ~mask; \ + } \ +} while (0) + +static struct davinci_pcm_dma_params davinci_vcif_pcm_out = { + .name = "VCIF PCM Stereo out", +}; + +static struct davinci_pcm_dma_params davinci_vcif_pcm_in = { + .name = "VCIF PCM Stereo in", +}; + +struct davinci_vcif_dev { + void __iomem *base; + resource_size_t pbase; + size_t base_size; + int mode; + u32 pcr; + struct clk *clk; + struct davinci_pcm_dma_params *dma_params[2]; +}; + +static inline void davinci_vcif_write_reg(struct davinci_vcif_dev *dev, + int reg, u32 val) +{ + __raw_writel(val, dev->base + reg); +} + +static inline u32 davinci_vcif_read_reg(struct davinci_vcif_dev *dev, int reg) +{ + return __raw_readl(dev->base + reg); +} + +static void davinci_vcif_start(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_vcif_dev *dev = rtd->dai->cpu_dai->private_data; + u32 w; + + /* Start the sample generator and enable transmitter/receiver */ + w = davinci_vcif_read_reg(dev, DAVINCI_VCIF_CTRL_REG); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + MOD_REG_BIT(w, DAVINCI_VCIF_CTRL_RSTDAC, 1); + else + MOD_REG_BIT(w, DAVINCI_VCIF_CTRL_RSTADC, 1); + + davinci_vcif_write_reg(dev, DAVINCI_VCIF_CTRL_REG, w); +} + +static void davinci_vcif_stop(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_vcif_dev *dev = rtd->dai->cpu_dai->private_data; + u32 w; + + /* Reset transmitter/receiver and sample rate/frame sync generators */ + w = davinci_vcif_read_reg(dev, DAVINCI_VCIF_CTRL_REG); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + MOD_REG_BIT(w, DAVINCI_VCIF_CTRL_RSTDAC, 0); + else + MOD_REG_BIT(w, DAVINCI_VCIF_CTRL_RSTADC, 0); + + davinci_vcif_write_reg(dev, DAVINCI_VCIF_CTRL_REG, w); +} + +static int davinci_vcif_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct davinci_vcif_dev *dev = cpu_dai->private_data; + + cpu_dai->dma_data = dev->dma_params[substream->stream]; + return 0; +} + +static int davinci_vcif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct davinci_pcm_dma_params *dma_params = dai->dma_data; + struct davinci_vcif_dev *dev = dai->private_data; + u32 w; + + /* Restart the codec before setup */ + davinci_vcif_stop(substream); + davinci_vcif_start(substream); + + /* General line settings */ + davinci_vcif_write_reg(dev, + DAVINCI_VCIF_CTRL_REG, DAVINCI_VCIF_CTRL_MASK); + + davinci_vcif_write_reg(dev, + DAVINCI_VCIF_INTCLR_REG, DAVINCI_VCIF_INT_MASK); + + davinci_vcif_write_reg(dev, + DAVINCI_VCIF_INTEN_REG, DAVINCI_VCIF_INT_MASK); + + w = davinci_vcif_read_reg(dev, DAVINCI_VCIF_CTRL_REG); + + /* Determine xfer data type */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U8: + dma_params->data_type = 0; + + MOD_REG_BIT(w, DAVINCI_VCIF_CTRL_RD_BITS_8 | + DAVINCI_VCIF_CTRL_RD_UNSIGNED | + DAVINCI_VCIF_CTRL_WD_BITS_8 | + DAVINCI_VCIF_CTRL_WD_UNSIGNED, 1); + break; + case SNDRV_PCM_FORMAT_S8: + dma_params->data_type = 1; + + MOD_REG_BIT(w, DAVINCI_VCIF_CTRL_RD_BITS_8 | + DAVINCI_VCIF_CTRL_WD_BITS_8, 1); + + MOD_REG_BIT(w, DAVINCI_VCIF_CTRL_RD_UNSIGNED | + DAVINCI_VCIF_CTRL_WD_UNSIGNED, 0); + break; + case SNDRV_PCM_FORMAT_S16_LE: + dma_params->data_type = 2; + + MOD_REG_BIT(w, DAVINCI_VCIF_CTRL_RD_BITS_8 | + DAVINCI_VCIF_CTRL_RD_UNSIGNED | + DAVINCI_VCIF_CTRL_WD_BITS_8 | + DAVINCI_VCIF_CTRL_WD_UNSIGNED, 0); + break; + default: + printk(KERN_WARNING "davinci-vcif: unsupported PCM format"); + return -EINVAL; + } + + dma_params->acnt = dma_params->data_type; + + davinci_vcif_write_reg(dev, DAVINCI_VCIF_CTRL_REG, w); + + return 0; +} + +static int davinci_vcif_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + davinci_vcif_start(substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + davinci_vcif_stop(substream); + break; + default: + ret = -EINVAL; + } + + return ret; +} + +#define DAVINCI_VCIF_RATES SNDRV_PCM_RATE_8000_48000 + +static struct snd_soc_dai_ops davinci_vcif_dai_ops = { + .startup = davinci_vcif_startup, + .trigger = davinci_vcif_trigger, + .hw_params = davinci_vcif_hw_params, +}; + +struct snd_soc_dai davinci_vcif_dai = { + .name = "davinci-vcif", + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = DAVINCI_VCIF_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = DAVINCI_VCIF_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &davinci_vcif_dai_ops, + +}; +EXPORT_SYMBOL_GPL(davinci_vcif_dai); + +static int davinci_vcif_probe(struct platform_device *pdev) +{ + struct davinci_vcif_dev *dev; + struct resource *res, *mem; + int ret; + + dev = kzalloc(sizeof(struct davinci_vcif_dev), GFP_KERNEL); + if (!dev) { + dev_dbg(&pdev->dev, "could not allocate memory for private data\n"); + return -ENOMEM; + } + + dev->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(dev->clk)) { + dev_dbg(&pdev->dev, "%s: could not get the clock for voice codec\n", + pdev->name); + ret = -ENODEV; + goto fail1; + } + clk_enable(dev->clk); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "no mem resource\n"); + ret = -ENODEV; + goto fail2; + } + + dev->pbase = res->start; + dev->base_size = resource_size(res); + + mem = request_mem_region(dev->pbase, dev->base_size, pdev->name); + if (!mem) { + dev_err(&pdev->dev, "VCIF region already claimed\n"); + ret = -EBUSY; + goto fail2; + } + + dev->base = ioremap(dev->pbase, dev->base_size); + if (!dev->base) { + dev_err(&pdev->dev,"%s: can't ioremap mem resource.\n", pdev->name); + ret = -ENOMEM; + goto fail3; + } + + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK] = &davinci_vcif_pcm_out; + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->dma_addr = + (dma_addr_t)(io_v2p(dev->base) + DAVINCI_VCIF_WFIFO_REG); + + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE] = &davinci_vcif_pcm_in; + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->dma_addr = + (dma_addr_t)(io_v2p(dev->base) + DAVINCI_VCIF_RFIFO_REG); + + /* first TX, then RX */ + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!res) { + dev_err(&pdev->dev, "%s: no DMA resource\n", pdev->name); + ret = -ENXIO; + goto fail4; + } + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->channel = res->start; + + res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!res) { + dev_err(&pdev->dev, "%s: no DMA resource\n", pdev->name); + ret = -ENXIO; + goto fail4; + } + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->channel = res->start; + + davinci_vcif_dai.private_data = dev; + ret = snd_soc_register_dai(&davinci_vcif_dai); + if (ret != 0) + goto fail4; + + return 0; + +fail4: + iounmap(dev->base); +fail3: + release_mem_region(dev->pbase, dev->base_size); +fail2: + clk_disable(dev->clk); + clk_put(dev->clk); + dev->clk = NULL; +fail1: + kfree(dev); + + return ret; +} + +static int davinci_vcif_remove(struct platform_device *pdev) +{ + struct davinci_vcif_dev *dev = davinci_vcif_dai.private_data; + + iounmap(dev->base); + release_mem_region(dev->pbase, dev->base_size); + + clk_disable(dev->clk); + clk_put(dev->clk); + dev->clk = NULL; + + kfree(dev); + + snd_soc_unregister_dai(&davinci_vcif_dai); + + return 0; +} + +static struct platform_driver davinci_vcif_driver = { + .probe = davinci_vcif_probe, + .remove = davinci_vcif_remove, + .driver = { + .name = "voice_codec", + .owner = THIS_MODULE, + }, +}; + +static int __init davinci_vcif_init(void) +{ + return platform_driver_probe(&davinci_vcif_driver, davinci_vcif_probe); +} +module_init(davinci_vcif_init); + +static void __exit davinci_vcif_exit(void) +{ + platform_driver_unregister(&davinci_vcif_driver); +} +module_exit(davinci_vcif_exit); + +MODULE_AUTHOR("Miguel Aguilar"); +MODULE_DESCRIPTION("Texas Instruments DaVinci ASoC Voice Codec Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/davinci/davinci-vcif.h b/sound/soc/davinci/davinci-vcif.h new file mode 100644 index 0000000..30e441d --- /dev/null +++ b/sound/soc/davinci/davinci-vcif.h @@ -0,0 +1,63 @@ +/* + * ALSA SoC Voice Codec Interface for TI DAVINCI processor + * + * Copyright (C) 2009 Texas Instruments. + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * Initial code: Hui Geng + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * 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 + */ + +#ifndef _DAVINCI_VCIF_H +#define _DAVINCI_VCIF_H + +#define DAVINCI_VCIF_PID_REG 0x00 +#define DAVINCI_VCIF_CTRL_REG 0x04 +#define DAVINCI_VCIF_INTEN_REG 0x08 +#define DAVINCI_VCIF_INTSTATUS_REG 0x0c +#define DAVINCI_VCIF_INTCLR_REG 0x10 +#define DAVINCI_VCIF_EMUL_CTRL_REG 0x14 +#define DAVINCI_VCIF_RFIFO_REG 0x20 +#define DAVINCI_VCIF_WFIFO_REG 0x24 +#define DAVINCI_VCIF_FIFOSTAT_REG 0x28 +#define DAVINCI_VCIF_TST_CTRL_REG 0x2C + +#define DAVINCI_VCIF_CTRL_MASK 0x5500 +#define DAVINCI_VCIF_CTRL_RSTADC (1 << 0) +#define DAVINCI_VCIF_CTRL_RSTDAC (1 << 1) +#define DAVINCI_VCIF_CTRL_RD_BITS_8 (1 << 4) +#define DAVINCI_VCIF_CTRL_RD_UNSIGNED (1 << 5) +#define DAVINCI_VCIF_CTRL_WD_BITS_8 (1 << 6) +#define DAVINCI_VCIF_CTRL_WD_UNSIGNED (1 << 7) +#define DAVINCI_VCIF_CTRL_RFIFOEN (1 << 8) +#define DAVINCI_VCIF_CTRL_RFIFOCL (1 << 9) +#define DAVINCI_VCIF_CTRL_RFIFOMD_WORD_1 (1 << 10) +#define DAVINCI_VCIF_CTRL_WFIFOEN (1 << 12) +#define DAVINCI_VCIF_CTRL_WFIFOCL (1 << 13) +#define DAVINCI_VCIF_CTRL_WFIFOMD_WORD_1 (1 << 14) + +#define DAVINCI_VCIF_INT_MASK 0x3F +#define DAVINCI_VCIF_INT_RDRDY_MASK (1 << 0) +#define DAVINCI_VCIF_INT_RERROVF_MASK (1 << 1) +#define DAVINCI_VCIF_INT_RERRUDR_MASK (1 << 2) +#define DAVINCI_VCIF_INT_WDREQ_MASK (1 << 3) +#define DAVINCI_VCIF_INT_WERROVF_MASK (1 << 4) +#define DAVINCI_VCIF_INT_WERRUDR_MASK (1 << 5) + +extern struct snd_soc_dai davinci_vcif_dai; + +#endif -- 1.6.0.4

3 7

Re: [alsa-devel] Testers wanted: New SB X-Fi driver
by Christian Esken 18 Nov '09

18 Nov '09

Am Sonntag 12 Juli 2009 10:24:35 schrieb Christian Esken: > Am Montag 06 Juli 2009 08:26:11 schrieb Takashi Iwai: > > At Sun, 5 Jul 2009 13:29:48 +0200, > > > > Christian Esken wrote: > > > Am Sonntag 05 Juli 2009 09:44:33 schrieb Takashi Iwai: > > > > At Sun, 5 Jul 2009 00:19:55 +0200, > > > > > > > > Christian Esken wrote: > > > > > Am Samstag 27 Juni 2009 14:05:45 schrieb Christian Esken: > > > > > > Hello, > > > > > > > > > > > > unfortunately I cannot playback anything (PCM) with the new SB > > > > > > X-Fi driver. Driver: > > > > > > ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa- > > > > > > driver-snapshot.tar.bz2 > > > > > > Distribution: openSuSE 11.1 > > > > > > Kernel: 2.6.27.23-0.1-xen > > > > > > > > > > > > My card is a "X-Fi Titanium" (PCI device code: 1102:000b). The > > > > > > card is recognized by the driver, and any mixer (alsamixer, kmix) > > > > > > also recognizes the card. > > > > > > > > > > > > But playing back isn't possible. I tried speaker-test, audacity, > > > > > > amarok and aplay. Most applications hang, some also produce an > > > > > > error in the /var/log/messages (see > > > > > > below). > > > > > > > > > > > > I then used "./configure --with-debug=full --with-cards=ctxfi" > > > > > > due to the possible hd-audio conflict, but it didn't make a > > > > > > difference. > > > > > > > > > > > > Is there anything else I could try? > > > > > > > > > > I tested the same openSuSE version with a non-Xen 64 bit kernel, > > > > > and it doesn't work either. > > > > > > > > > > And I tried with this: > > > > > Driver: alsa-driver-20090704.tar.bz2 > > > > > Distribution: Kubuntu Karmic Alpha-2 > > > > > Kernel: 2.6.30 (32 Bit) > > > > > > > > > > Under the latter setup the X-Fi Titanium works (Simple 2 channel > > > > > test with speaker-test, audacity, amarok). :-)) > > > > > > > > How many RAM do you have? > > > > Could you boot with the limited memory size? > > > > > > It is just 2GB. So I am not limited by the 32 bit barrier. > > > > > > So booting with "limited" memory size doesn't make sense, right? > > > > Supposedly. Is it emu20k1 or emu20k2? > > I checked it again: It is a 20k2 chip. > > /var/log/boot.msg says: > <7>ALSA /home/chris/Desktop/alsa-driver/pci/ctxfi/ctatc.c:1268: ctxfi: chip > 20K2 model SB0880 (1102:0041) is found > > > http://pci-ids.ucw.cz/mods/PC/1102/000b shows 1102:0041 as "X-Fi Titanium > series [EMU20k2]". I did now test it with a different 64 Bit Linux distribution, and it works correctly: Driver: alsa-driver-20090712.tar.bz2 Distribution: Kubuntu Karmic Alpha-2 Kernel: 2.6.30 (64 Bit) All in all, I can produce a working environment with recent 32 and 64 bit kernels. This is good news. :-) Below is the summary on the tested distributions. Christian So we got this: Kubuntu Karmic Alpha-2 32 Bit, 2.0.30 (works) Kubuntu Karmic Alpha-2 64 Bit, 2.0.30 (works) SuSE 11.1 64 Bit, 2.0.27 (does not work) SuSE 11.1 64 Bit, 2.0.27 XEN (does not work) SuSE 11.2 Milestone3 64Bit, 2.0.30 (test canceled, due to unrelated problems) => I'll see whether I can redo the SuSE11.2 tests, after doing a bugreport at openSuSE.

3 6

[alsa-devel] [RFC][PATCH] ARM: OMAP: McBSP: Use register cache
by Janusz Krzysztofik 14 Nov '09

14 Nov '09

Change the way McBSP registers are updated: use cached values instead of relying upon those read back from the device. With this patch, I have finally managed to get rid of all random playback/recording hangups on my OMAP1510 based Amstrad Delta (buggy?) hardware. Before that, I could suspect that values read back from McBSP registers before updating them happened to be errornous. I think there is one important point that makes this patch worth of applying, apart from my hardware quality. With the current code, if it ever happens to any machine, no matter if OMAP1510 or newer, to read incorrect value from a McBSP register, this wrong value will get written back without any checking. That can lead to hardware damage if, for example, an input pin is turned into output as a result. Created against linux-2.6.31-rc5 Tested on Amstrad Delta Signed-off-by: Janusz Krzysztofik --- --- linux-2.6.31-rc5/arch/arm/plat-omap/include/mach/mcbsp.h.orig 2009-08-11 23:43:25.000000000 +0200 +++ linux-2.6.31-rc5/arch/arm/plat-omap/include/mach/mcbsp.h 2009-08-11 23:45:46.000000000 +0200 @@ -377,6 +377,8 @@ struct omap_mcbsp { struct omap_mcbsp_platform_data *pdata; struct clk *iclk; struct clk *fclk; + + struct omap_mcbsp_reg_cfg reg_cache; }; extern struct omap_mcbsp **mcbsp_ptr; extern int omap_mcbsp_count; --- linux-2.6.31-rc5/arch/arm/plat-omap/mcbsp.c.orig 2009-08-11 23:43:25.000000000 +0200 +++ linux-2.6.31-rc5/arch/arm/plat-omap/mcbsp.c 2009-08-11 23:45:35.000000000 +0200 @@ -91,6 +91,7 @@ static void omap_mcbsp_dump_reg(u8 id) static irqreturn_t omap_mcbsp_tx_irq_handler(int irq, void *dev_id) { struct omap_mcbsp *mcbsp_tx = dev_id; + struct omap_mcbsp_reg_cfg *reg_cache = &mcbsp_tx->reg_cache; u16 irqst_spcr2; irqst_spcr2 = OMAP_MCBSP_READ(mcbsp_tx->io_base, SPCR2); @@ -101,7 +102,7 @@ static irqreturn_t omap_mcbsp_tx_irq_han irqst_spcr2); /* Writing zero to XSYNC_ERR clears the IRQ */ OMAP_MCBSP_WRITE(mcbsp_tx->io_base, SPCR2, - irqst_spcr2 & ~(XSYNC_ERR)); + reg_cache->spcr2 &= ~(XSYNC_ERR)); } else { complete(&mcbsp_tx->tx_irq_completion); } @@ -112,6 +113,7 @@ static irqreturn_t omap_mcbsp_tx_irq_han static irqreturn_t omap_mcbsp_rx_irq_handler(int irq, void *dev_id) { struct omap_mcbsp *mcbsp_rx = dev_id; + struct omap_mcbsp_reg_cfg *reg_cache = &mcbsp_rx->reg_cache; u16 irqst_spcr1; irqst_spcr1 = OMAP_MCBSP_READ(mcbsp_rx->io_base, SPCR1); @@ -122,7 +124,7 @@ static irqreturn_t omap_mcbsp_rx_irq_han irqst_spcr1); /* Writing zero to RSYNC_ERR clears the IRQ */ OMAP_MCBSP_WRITE(mcbsp_rx->io_base, SPCR1, - irqst_spcr1 & ~(RSYNC_ERR)); + reg_cache->spcr1 &= ~(RSYNC_ERR)); } else { complete(&mcbsp_rx->tx_irq_completion); } @@ -167,6 +169,7 @@ static void omap_mcbsp_rx_dma_callback(i void omap_mcbsp_config(unsigned int id, const struct omap_mcbsp_reg_cfg *config) { struct omap_mcbsp *mcbsp; + struct omap_mcbsp_reg_cfg *reg_cache; void __iomem *io_base; if (!omap_mcbsp_check_valid_id(id)) { @@ -174,26 +177,27 @@ void omap_mcbsp_config(unsigned int id, return; } mcbsp = id_to_mcbsp_ptr(id); + reg_cache = &mcbsp->reg_cache; io_base = mcbsp->io_base; dev_dbg(mcbsp->dev, "Configuring McBSP%d phys_base: 0x%08lx\n", mcbsp->id, mcbsp->phys_base); /* We write the given config */ - OMAP_MCBSP_WRITE(io_base, SPCR2, config->spcr2); - OMAP_MCBSP_WRITE(io_base, SPCR1, config->spcr1); - OMAP_MCBSP_WRITE(io_base, RCR2, config->rcr2); - OMAP_MCBSP_WRITE(io_base, RCR1, config->rcr1); - OMAP_MCBSP_WRITE(io_base, XCR2, config->xcr2); - OMAP_MCBSP_WRITE(io_base, XCR1, config->xcr1); - OMAP_MCBSP_WRITE(io_base, SRGR2, config->srgr2); - OMAP_MCBSP_WRITE(io_base, SRGR1, config->srgr1); - OMAP_MCBSP_WRITE(io_base, MCR2, config->mcr2); - OMAP_MCBSP_WRITE(io_base, MCR1, config->mcr1); - OMAP_MCBSP_WRITE(io_base, PCR0, config->pcr0); + OMAP_MCBSP_WRITE(io_base, SPCR2, reg_cache->spcr2 = config->spcr2); + OMAP_MCBSP_WRITE(io_base, SPCR1, reg_cache->spcr1 = config->spcr1); + OMAP_MCBSP_WRITE(io_base, RCR2, reg_cache->rcr2 = config->rcr2); + OMAP_MCBSP_WRITE(io_base, RCR1, reg_cache->rcr1 = config->rcr1); + OMAP_MCBSP_WRITE(io_base, XCR2, reg_cache->xcr2 = config->xcr2); + OMAP_MCBSP_WRITE(io_base, XCR1, reg_cache->xcr1 = config->xcr1); + OMAP_MCBSP_WRITE(io_base, SRGR2, reg_cache->srgr2 = config->srgr2); + OMAP_MCBSP_WRITE(io_base, SRGR1, reg_cache->srgr1 = config->srgr1); + OMAP_MCBSP_WRITE(io_base, MCR2, reg_cache->mcr2 = config->mcr2); + OMAP_MCBSP_WRITE(io_base, MCR1, reg_cache->mcr1 = config->mcr1); + OMAP_MCBSP_WRITE(io_base, PCR0, reg_cache->pcr0 = config->pcr0); if (cpu_is_omap2430() || cpu_is_omap34xx()) { - OMAP_MCBSP_WRITE(io_base, XCCR, config->xccr); - OMAP_MCBSP_WRITE(io_base, RCCR, config->rccr); + OMAP_MCBSP_WRITE(io_base, XCCR, reg_cache->xccr = config->xccr); + OMAP_MCBSP_WRITE(io_base, RCCR, reg_cache->rccr = config->rccr); } } EXPORT_SYMBOL(omap_mcbsp_config); @@ -232,6 +236,7 @@ EXPORT_SYMBOL(omap_mcbsp_set_io_type); int omap_mcbsp_request(unsigned int id) { struct omap_mcbsp *mcbsp; + struct omap_mcbsp_reg_cfg *reg_cache; int err; if (!omap_mcbsp_check_valid_id(id)) { @@ -239,6 +244,7 @@ int omap_mcbsp_request(unsigned int id) return -ENODEV; } mcbsp = id_to_mcbsp_ptr(id); + reg_cache = &mcbsp->reg_cache; spin_lock(&mcbsp->lock); if (!mcbsp->free) { @@ -261,8 +267,8 @@ int omap_mcbsp_request(unsigned int id) * Make sure that transmitter, receiver and sample-rate generator are * not running before activating IRQs. */ - OMAP_MCBSP_WRITE(mcbsp->io_base, SPCR1, 0); - OMAP_MCBSP_WRITE(mcbsp->io_base, SPCR2, 0); + OMAP_MCBSP_WRITE(mcbsp->io_base, SPCR1, reg_cache->spcr1 = 0); + OMAP_MCBSP_WRITE(mcbsp->io_base, SPCR2, reg_cache->spcr2 = 0); if (mcbsp->io_type == OMAP_MCBSP_IRQ_IO) { /* We need to get IRQs here */ @@ -335,42 +341,38 @@ EXPORT_SYMBOL(omap_mcbsp_free); void omap_mcbsp_start(unsigned int id, int tx, int rx) { struct omap_mcbsp *mcbsp; + struct omap_mcbsp_reg_cfg *reg_cache; void __iomem *io_base; int idle; - u16 w; if (!omap_mcbsp_check_valid_id(id)) { printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); return; } mcbsp = id_to_mcbsp_ptr(id); + reg_cache = &mcbsp->reg_cache; io_base = mcbsp->io_base; - mcbsp->rx_word_length = (OMAP_MCBSP_READ(io_base, RCR1) >> 5) & 0x7; - mcbsp->tx_word_length = (OMAP_MCBSP_READ(io_base, XCR1) >> 5) & 0x7; + mcbsp->rx_word_length = (reg_cache->rcr1 >> 5) & 0x7; + mcbsp->tx_word_length = (reg_cache->xcr1 >> 5) & 0x7; - idle = !((OMAP_MCBSP_READ(io_base, SPCR2) | - OMAP_MCBSP_READ(io_base, SPCR1)) & 1); + idle = !((reg_cache->spcr2 | reg_cache->spcr1) & 1); if (idle) { /* Start the sample generator */ - w = OMAP_MCBSP_READ(io_base, SPCR2); - OMAP_MCBSP_WRITE(io_base, SPCR2, w | (1 << 6)); + OMAP_MCBSP_WRITE(io_base, SPCR2, reg_cache->spcr2 |= (1 << 6)); } /* Enable transmitter and receiver */ - w = OMAP_MCBSP_READ(io_base, SPCR2); - OMAP_MCBSP_WRITE(io_base, SPCR2, w | (tx & 1)); + OMAP_MCBSP_WRITE(io_base, SPCR2, reg_cache->spcr2 |= (tx & 1)); - w = OMAP_MCBSP_READ(io_base, SPCR1); - OMAP_MCBSP_WRITE(io_base, SPCR1, w | (rx & 1)); + OMAP_MCBSP_WRITE(io_base, SPCR1, reg_cache->spcr1 |= (rx & 1)); udelay(100); if (idle) { /* Start frame sync */ - w = OMAP_MCBSP_READ(io_base, SPCR2); - OMAP_MCBSP_WRITE(io_base, SPCR2, w | (1 << 7)); + OMAP_MCBSP_WRITE(io_base, SPCR2, reg_cache->spcr2 |= (1 << 7)); } /* Dump McBSP Regs */ @@ -381,9 +383,9 @@ EXPORT_SYMBOL(omap_mcbsp_start); void omap_mcbsp_stop(unsigned int id, int tx, int rx) { struct omap_mcbsp *mcbsp; + struct omap_mcbsp_reg_cfg *reg_cache; void __iomem *io_base; int idle; - u16 w; if (!omap_mcbsp_check_valid_id(id)) { printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); @@ -391,23 +393,20 @@ void omap_mcbsp_stop(unsigned int id, in } mcbsp = id_to_mcbsp_ptr(id); + reg_cache = &mcbsp->reg_cache; io_base = mcbsp->io_base; /* Reset transmitter */ - w = OMAP_MCBSP_READ(io_base, SPCR2); - OMAP_MCBSP_WRITE(io_base, SPCR2, w & ~(tx & 1)); + OMAP_MCBSP_WRITE(io_base, SPCR2, reg_cache->spcr2 &= ~(tx & 1)); /* Reset receiver */ - w = OMAP_MCBSP_READ(io_base, SPCR1); - OMAP_MCBSP_WRITE(io_base, SPCR1, w & ~(rx & 1)); + OMAP_MCBSP_WRITE(io_base, SPCR1, reg_cache->spcr1 &= ~(rx & 1)); - idle = !((OMAP_MCBSP_READ(io_base, SPCR2) | - OMAP_MCBSP_READ(io_base, SPCR1)) & 1); + idle = !((reg_cache->spcr2 | reg_cache->spcr1) & 1); if (idle) { /* Reset the sample rate generator */ - w = OMAP_MCBSP_READ(io_base, SPCR2); - OMAP_MCBSP_WRITE(io_base, SPCR2, w & ~(1 << 6)); + OMAP_MCBSP_WRITE(io_base, SPCR2, reg_cache->spcr2 &= ~(1 << 6)); } } EXPORT_SYMBOL(omap_mcbsp_stop); @@ -557,6 +556,7 @@ EXPORT_SYMBOL(omap_mcbsp_recv_word); int omap_mcbsp_spi_master_xmit_word_poll(unsigned int id, u32 word) { struct omap_mcbsp *mcbsp; + struct omap_mcbsp_reg_cfg *reg_cache; void __iomem *io_base; omap_mcbsp_word_length tx_word_length; omap_mcbsp_word_length rx_word_length; @@ -567,6 +567,7 @@ int omap_mcbsp_spi_master_xmit_word_poll return -ENODEV; } mcbsp = id_to_mcbsp_ptr(id); + reg_cache = &mcbsp->reg_cache; io_base = mcbsp->io_base; tx_word_length = mcbsp->tx_word_length; rx_word_length = mcbsp->rx_word_length; @@ -580,9 +581,11 @@ int omap_mcbsp_spi_master_xmit_word_poll spcr2 = OMAP_MCBSP_READ(io_base, SPCR2); if (attempts++ > 1000) { /* We must reset the transmitter */ - OMAP_MCBSP_WRITE(io_base, SPCR2, spcr2 & (~XRST)); + OMAP_MCBSP_WRITE(io_base, SPCR2, + reg_cache->spcr2 &= (~XRST)); udelay(10); - OMAP_MCBSP_WRITE(io_base, SPCR2, spcr2 | XRST); + OMAP_MCBSP_WRITE(io_base, SPCR2, + reg_cache->spcr2 |= XRST); udelay(10); dev_err(mcbsp->dev, "McBSP%d transmitter not " "ready\n", mcbsp->id); @@ -601,9 +604,11 @@ int omap_mcbsp_spi_master_xmit_word_poll spcr1 = OMAP_MCBSP_READ(io_base, SPCR1); if (attempts++ > 1000) { /* We must reset the receiver */ - OMAP_MCBSP_WRITE(io_base, SPCR1, spcr1 & (~RRST)); + OMAP_MCBSP_WRITE(io_base, SPCR1, + reg_cache->spcr1 &= (~RRST)); udelay(10); - OMAP_MCBSP_WRITE(io_base, SPCR1, spcr1 | RRST); + OMAP_MCBSP_WRITE(io_base, SPCR1, + reg_cache->spcr1 |= RRST); udelay(10); dev_err(mcbsp->dev, "McBSP%d receiver not " "ready\n", mcbsp->id); @@ -623,6 +628,7 @@ EXPORT_SYMBOL(omap_mcbsp_spi_master_xmit int omap_mcbsp_spi_master_recv_word_poll(unsigned int id, u32 *word) { struct omap_mcbsp *mcbsp; + struct omap_mcbsp_reg_cfg *reg_cache; u32 clock_word = 0; void __iomem *io_base; omap_mcbsp_word_length tx_word_length; @@ -635,6 +641,7 @@ int omap_mcbsp_spi_master_recv_word_poll } mcbsp = id_to_mcbsp_ptr(id); + reg_cache = &mcbsp->reg_cache; io_base = mcbsp->io_base; tx_word_length = mcbsp->tx_word_length; @@ -649,9 +656,11 @@ int omap_mcbsp_spi_master_recv_word_poll spcr2 = OMAP_MCBSP_READ(io_base, SPCR2); if (attempts++ > 1000) { /* We must reset the transmitter */ - OMAP_MCBSP_WRITE(io_base, SPCR2, spcr2 & (~XRST)); + OMAP_MCBSP_WRITE(io_base, SPCR2, + reg_cache->spcr2 &= (~XRST)); udelay(10); - OMAP_MCBSP_WRITE(io_base, SPCR2, spcr2 | XRST); + OMAP_MCBSP_WRITE(io_base, SPCR2, + reg_cache->spcr2 |= XRST); udelay(10); dev_err(mcbsp->dev, "McBSP%d transmitter not " "ready\n", mcbsp->id); @@ -670,9 +679,11 @@ int omap_mcbsp_spi_master_recv_word_poll spcr1 = OMAP_MCBSP_READ(io_base, SPCR1); if (attempts++ > 1000) { /* We must reset the receiver */ - OMAP_MCBSP_WRITE(io_base, SPCR1, spcr1 & (~RRST)); + OMAP_MCBSP_WRITE(io_base, SPCR1, + reg_cache->spcr1 &= (~RRST)); udelay(10); - OMAP_MCBSP_WRITE(io_base, SPCR1, spcr1 | RRST); + OMAP_MCBSP_WRITE(io_base, SPCR1, + reg_cache->spcr1 |= RRST); udelay(10); dev_err(mcbsp->dev, "McBSP%d receiver not " "ready\n", mcbsp->id);

1 2

[alsa-devel] Trying to get unsupported CODEC to work
by Alex Austin 05 Nov '09

05 Nov '09

HelloI have an MSI Media Live DIVA motherboard. It has an AMD SB600 HD Audio controller, and two CODECs on the HDA bus. First is an ALC888 Second is an Intersil DAE-3. Two alsa-info outputs in different circumstances http://www.alsa-project.org/db/?f=20c86e755fdbbdbc8761c8c2f4de76bce2ea0c11 http://www.alsa-project.org/db/?f=15c5e5eab0104622a84b2dd0df03f26c17c7242d Output of codecgraph with "modprobe hda-intel probe_mask=2" http://pastebin.com/f54b6a9b8 What's the next step to get this supported? -- - Alex Austin (651) 238-9273 "...and then I visited Wikipedia ...and the next 8 hours are a blur."

3 13

[alsa-devel] [Fwd: Scratch Live amplifier switch]
by LCID Fire 02 Nov '09

02 Nov '09

Hi. Following the patch to add the amplifier switch (changing input levels phono/line) for the Serato Scratch Live device. Please apply or complain ;)

2 3

[alsa-devel] [PATCH] ASoC: OMAP3EVM: Use the twl4030_setup_data for headset pop-removal
by Anuj Aggarwal 29 Oct '09

29 Oct '09

The pop-removal specific values are configured for TWL4030 codec for OMAP3EVM through this patch. Signed-off-by: Anuj Aggarwal <anuj.aggarwal(a)ti.com> --- sound/soc/omap/omap3evm.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c index 9114c26..8deb59b 100644 --- a/sound/soc/omap/omap3evm.c +++ b/sound/soc/omap/omap3evm.c @@ -93,10 +93,17 @@ static struct snd_soc_card snd_soc_omap3evm = { .num_links = 1, }; +/* twl4030 setup */ +static struct twl4030_setup_data twl4030_setup = { + .ramp_delay_value = 4, + .sysclk = 26000, +}; + /* Audio subsystem */ static struct snd_soc_device omap3evm_snd_devdata = { .card = &snd_soc_omap3evm, .codec_dev = &soc_codec_dev_twl4030, + .codec_data = &twl4030_setup, }; static struct platform_device *omap3evm_snd_device; -- 1.6.2.4

4 3

[alsa-devel] Questions on locking, snd-usb-caiaq
by Mark Hills 18 Oct '09

18 Oct '09

I fixed some race issues in snd-usb-caiaq, but have some questions on the interface between the driver and ALSA. First fix is a lock in pointer(), which caused snd_pcm_update_hw_ptr_pos() to report an out of range buffer position (even though it was sucessfully corrected to zero). Second is a lock as part of trigger(). On testing, this appears to fix an issue where white noise could sometimes be transferred by the driver. According to the documentation, trigger() and pointer() are called with local interrupts disabled. This means they should really be non-irqsave locks? (http://www.alsa-project.org/~tiwai/writing-an-alsa-driver/ch05s08.html) Even for callbacks documented as atomic, if read_completed() and write_completed() can be called at any time interrupts are enabled (eg. on another CPU) it seems that additional locking my be required. Candidates are: snd_usb_caiaq_substream_open snd_usb_caiaq_substream_close snd_usb_caiaq_pcm_hw_free snd_usb_caiaq_pcm_prepare Are there any guarantees on these functions which mean the lock is not required? To lock in prepare() would (I think) require demoting a spin_lock_irqsave() to a spin_lock() to send initialisation commands. If so, what is the best design for this? Thanks, -- Mark diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 121af06..c659f81 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -62,10 +62,15 @@ static void activate_substream(struct snd_usb_caiaqdev *dev, struct snd_pcm_substream *sub) { + unsigned long flags; + spin_lock_irqsave(&dev->spinlock, flags); + if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK) dev->sub_playback[sub->number] = sub; else dev->sub_capture[sub->number] = sub; + + spin_unlock_irqrestore(&dev->spinlock, flags); } static void @@ -269,16 +274,23 @@ snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub) { int index = sub->number; struct snd_usb_caiaqdev *dev = snd_pcm_substream_chip(sub); + unsigned long flags; + snd_pcm_uframes_t ptr; + + spin_lock_irqsave(&dev->spinlock, flags); if (dev->input_panic || dev->output_panic) - return SNDRV_PCM_POS_XRUN; + ptr = SNDRV_PCM_POS_XRUN; if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK) - return bytes_to_frames(sub->runtime, + ptr = bytes_to_frames(sub->runtime, dev->audio_out_buf_pos[index]); else - return bytes_to_frames(sub->runtime, + ptr = bytes_to_frames(sub->runtime, dev->audio_in_buf_pos[index]); + + spin_unlock_irqrestore(&dev->spinlock, flags); + return ptr; } /* operators for both playback and capture */

3 12

[alsa-devel] [PATCHv2 5/7] ASoC: TWL6030: Add support for low-power mode
by Lopez Cruz, Misael 14 Oct '09

14 Oct '09

TWL6030 codec supports two power modes: low-power and high performance. In low-power mode, headset downlink must be the only path enabled and components in that path (headset DAC and driver) should be in that mode too. In this mode, codec can stream audio at 44.1 and 48 kHz if sys clock from CLK32K is configured to 17.64 and 19.2 MHz using low-power PLL, respectively. In high-performance mode, codec can only work at 19.2 MHz from high-performance PLL. All paths supported in the codec can be used but audio can be streamed only at 48 kHz. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl6030.c | 214 ++++++++++++++++++++++++++++++++++++++------ sound/soc/codecs/twl6030.h | 17 ++++ 2 files changed, 202 insertions(+), 29 deletions(-) diff --git a/sound/soc/codecs/twl6030.c b/sound/soc/codecs/twl6030.c index f1e333f..032619d 100644 --- a/sound/soc/codecs/twl6030.c +++ b/sound/soc/codecs/twl6030.c @@ -46,6 +46,7 @@ struct twl6030_data { struct snd_soc_codec codec; int codec_powered; + unsigned int sysclk; }; /* @@ -305,6 +306,29 @@ static void twl6030_power_down(struct snd_soc_codec *codec) mdelay(10); } +/* set headset dac and driver power mode */ +static int headset_power_mode(struct snd_soc_codec *codec, int high_perf) +{ + int hslctl, hsrctl; + int mask = TWL6030_HSDRVMODEL | TWL6030_HSDACMODEL; + + hslctl = twl6030_read_reg_cache(codec, TWL6030_REG_HSLCTL); + hsrctl = twl6030_read_reg_cache(codec, TWL6030_REG_HSRCTL); + + if (high_perf) { + hslctl &= ~mask; + hsrctl &= ~mask; + } else { + hslctl |= mask; + hsrctl |= mask; + } + + twl6030_write(codec, TWL6030_REG_HSLCTL, hslctl); + twl6030_write(codec, TWL6030_REG_HSRCTL, hsrctl); + + return 0; +} + /* * MICATT volume control: * from -6 to 0 dB in 6 dB steps @@ -598,6 +622,7 @@ static int twl6030_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; + struct twl6030_data *priv = codec->private_data; int rate, format; /* hardware dai (McPDM) requires bit stream of twice @@ -607,7 +632,20 @@ static int twl6030_hw_params(struct snd_pcm_substream *substream, rate = params_rate(params); switch (rate) { case 44100: + if (priv->sysclk != 17640000) { + dev_err(codec->dev, + "rate %d not supported at current sysclk %d\n", + rate, priv->sysclk); + return -EINVAL; + } + break; case 48000: + if (priv->sysclk != 19200000) { + dev_err(codec->dev, + "rate %d not supported at current sysclk %d\n", + rate, priv->sysclk); + return -EINVAL; + } break; default: dev_err(codec->dev, "unknown rate %d\n", rate); @@ -630,46 +668,163 @@ static int twl6030_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; + struct twl6030_data *priv = codec->private_data; u8 hppll, lppll; + lppll = twl6030_read_reg_cache(codec, TWL6030_REG_LPPLLCTL); hppll = twl6030_read_reg_cache(codec, TWL6030_REG_HPPLLCTL); - hppll &= TWL6030_HPLLRST; - - switch (freq) { - case 12000000: - /* MCLK input, PLL enabled */ - hppll = TWL6030_MCLK_12000KHZ - | TWL6030_HPLLSQRBP - | TWL6030_HPLLENA; - break; - case 19200000: - /* MCLK input, PLL disabled */ - hppll = TWL6030_MCLK_19200KHZ - | TWL6030_HPLLSQRBP - | TWL6030_HPLLBP; + + switch (clk_id) { + case TWL6030_SYSCLK_SEL_LPPLL: + if (freq != 32768) { + dev_err(codec->dev, "invalid sysclk freq %d\n", freq); + return -EINVAL; + } + + /* CLK32K input requires low-power PLL */ + lppll |= TWL6030_LPLLENA | TWL6030_LPLLSEL; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + mdelay(5); + lppll &= ~TWL6030_HPLLSEL; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + + /* headset dac and driver must be in low-power mode */ + headset_power_mode(codec, 0); + + hppll &= ~TWL6030_HPLLENA; + twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppll); break; - case 26000000: - /* MCLK input, PLL enabled */ - hppll = TWL6030_MCLK_26000KHZ - | TWL6030_HPLLSQRBP - | TWL6030_HPLLENA; + case TWL6030_SYSCLK_SEL_HPPLL: + switch (freq) { + case 12000000: + hppll = TWL6030_MCLK_12000KHZ; + break; + case 26000000: + hppll = TWL6030_MCLK_26000KHZ; + break; + default: + dev_err(codec->dev, "invalid sysclk freq %d\n", freq); + return -EINVAL; + } + + /* 12 and 26 MHz freqs require high-performance PLL */ + hppll |= TWL6030_HPLLSQRBP | TWL6030_HPLLENA; + twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppll); + udelay(500); + + /* headset dac and driver must be in high-performance mode */ + headset_power_mode(codec, 1); + + lppll |= TWL6030_HPLLSEL; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + lppll &= ~TWL6030_LPLLENA; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + + /* only 19.2 MHz can be generated by HPPLL */ + priv->sysclk = 19200000; break; - case 38400000: - /* clk slicer input, PLL disabled */ - hppll = TWL6030_MCLK_38400KHZ - | TWL6030_HPLLSQRENA - | TWL6030_HPLLBP; + case TWL6030_SYSCLK_SEL_MCLK: + switch (freq) { + case 19200000: + hppll = TWL6030_MCLK_19200KHZ | TWL6030_HPLLSQRBP; + break; + case 38400000: + hppll = TWL6030_MCLK_38400KHZ; + break; + default: + dev_err(codec->dev, "invalid sysclk freq %d\n", freq); + return -EINVAL; + } + + /* 19.2 and 38.4 MHz freqs don't require PLL */ + hppll |= TWL6030_HPLLBP; + twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppll); + udelay(500); + + /* headset dac and driver must be in high-performance mode */ + headset_power_mode(codec, 1); + + lppll |= TWL6030_HPLLSEL; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + lppll &= ~TWL6030_LPLLENA; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + + /* only 19.2 MHz can be generated by MCLK */ + priv->sysclk = 19200000; break; default: - dev_err(codec->dev, "unknown sysclk rate %d\n", freq); + dev_err(codec->dev, "unknown clk_id %d\n", clk_id); return -EINVAL; } - twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppll); + return 0; +} + +static int twl6030_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct twl6030_data *priv = codec->private_data; + int div; + u8 ldoctl, lppllctl, hppllctl; + + ldoctl = twl6030_read_reg_cache(codec, TWL6030_REG_LDOCTL); + lppllctl = twl6030_read_reg_cache(codec, TWL6030_REG_LPPLLCTL); + hppllctl = twl6030_read_reg_cache(codec, TWL6030_REG_HPPLLCTL); + + if (freq_in && freq_out) { + /* div = round(freq_out / freq_in) */ + div = (freq_out + (freq_in >> 2)) / freq_in; + + switch (pll_id) { + case TWL6030_LPPLL_ID: + if (div < 512) { + dev_err(codec->dev, "invalid pll div %d\n", + div); + return -EINVAL; + } + twl6030_write(codec, TWL6030_REG_LPPLLDIV, div - 512); + lppllctl |= TWL6030_LPLLENA; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppllctl); + + priv->sysclk = freq_out; + break; + case TWL6030_HPPLL_ID: + hppllctl |= TWL6030_HPLLENA; + twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppllctl); + break; + default: + dev_err(codec->dev, "unknown pll id %d\n", pll_id); + return -EINVAL; + } - /* Disable LPPLL and select HPPLL */ - lppll = TWL6030_HPLLSEL; - twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + /* switch to pll */ + ldoctl &= ~TWL6030_OSCENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + udelay(10); + } else { + /* switch to internal oscillator */ + ldoctl |= TWL6030_OSCENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + udelay(10); + + /* turn-off pll */ + switch (pll_id) { + case TWL6030_HPPLL_ID: + hppllctl &= ~TWL6030_HPLLENA; + twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppllctl); + break; + case TWL6030_LPPLL_ID: + lppllctl &= ~TWL6030_LPLLENA; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppllctl); + break; + default: + dev_err(codec->dev, "unknown pll id %d\n", pll_id); + return -EINVAL; + } + + priv->sysclk = 0; + } return 0; } @@ -689,6 +844,7 @@ static int twl6030_set_dai_fmt(struct snd_soc_dai *codec_dai, static struct snd_soc_dai_ops twl6030_dai_ops = { .hw_params = twl6030_hw_params, .set_sysclk = twl6030_set_dai_sysclk, + .set_pll = twl6030_set_dai_pll, .set_fmt = twl6030_set_dai_fmt, }; diff --git a/sound/soc/codecs/twl6030.h b/sound/soc/codecs/twl6030.h index 7375ae8..15d3e1b 100644 --- a/sound/soc/codecs/twl6030.h +++ b/sound/soc/codecs/twl6030.h @@ -92,6 +92,10 @@ #define TWL6030_MCLK_38400KHZ (3 << 5) #define TWL6030_MCLK_MSK 0x60 +#define TWL6030_SYSCLK_SEL_LPPLL 1 +#define TWL6030_SYSCLK_SEL_HPPLL 2 +#define TWL6030_SYSCLK_SEL_MCLK 3 + /* LPPLLCTL (0x08) fields */ #define TWL6030_LPLLENA 0x01 @@ -100,6 +104,19 @@ #define TWL6030_LPLLFIN 0x08 #define TWL6030_HPLLSEL 0x10 +#define TWL6030_HPPLL_ID 1 +#define TWL6030_LPPLL_ID 2 + +/* HSLCTL (0x10) fields */ + +#define TWL6030_HSDACMODEL 0x02 +#define TWL6030_HSDRVMODEL 0x08 + +/* HSRCTL (0x11) fields */ + +#define TWL6030_HSDACMODER 0x02 +#define TWL6030_HSDRVMODER 0x08 + /* ACCCTL (0x2D) fields */ #define TWL6030_RESETSPLIT 0x04 -- 1.5.4.3

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[alsa-devel] [PATCH] snd-usb-us122l: added support for US-144
by Tobias Hansen 13 Oct '09

13 Oct '09

Hi, here is a patch that adds support for the Tascam US-144 to snd-usb-us122l when uhci-hcd is used (ehci-hcd has to be disabled). With ehci-hcd the US-144 behaves completely differently. The difference between US-122L and US-144 on uhci-hcd is, that US-144 uses endpoints on two interfaces, while on US-122L all endpoints are on one interface. US-122L should still work, but I don't have a US-122L to test that. Audio in and out and Midi in were tested with US-144 and work. This is my first contribution to the linux kernel, so please have a critical look on it. ;) Taken against snd-usb-us122l 0.5 / alsa 1.0.20. Regards, Tobias Hansen

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