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

[alsa-devel] [PATCH v2] ASoC: Tegra: Implement SPDIF CPU DAI
by Stephen Warren 05 Jul '11

05 Jul '11

This is a minimal driver for the Tegra SPDIF controller. In hardware, the SPDIF output signal is always routed to any active HDMI display controllers, and may also be routed to external pins on Tegra using the pinmux. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> Acked-by: Liam Girdwood <lrg(a)ti.com> --- v2: Fixed tegra_spdif_dai.name; removed trailing ".0". This worked previously because the spdif platform device's ID was -1, which caused fmt_single_name to throw away the "." and anything after it. sound/soc/tegra/Kconfig | 9 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_spdif.c | 371 ++++++++++++++++++++++++++++++++ sound/soc/tegra/tegra_spdif.h | 473 +++++++++++++++++++++++++++++++++++++++++ 4 files changed, 855 insertions(+), 0 deletions(-) create mode 100644 sound/soc/tegra/tegra_spdif.c create mode 100644 sound/soc/tegra/tegra_spdif.h diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 035d39a..c6af1fd 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -12,6 +12,15 @@ config SND_SOC_TEGRA_I2S Tegra I2S interface. You will also need to select the individual machine drivers to support below. +config SND_SOC_TEGRA_SPDIF + tristate + depends on SND_SOC_TEGRA + default m + help + Say Y or M if you want to add support for the SPDIF interface. + You will also need to select the individual machine drivers to support + below. + config MACH_HAS_SND_SOC_TEGRA_WM8903 bool help diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index fa6574d..4d943b3 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -2,12 +2,14 @@ snd-soc-tegra-das-objs := tegra_das.o snd-soc-tegra-pcm-objs := tegra_pcm.o snd-soc-tegra-i2s-objs := tegra_i2s.o +snd-soc-tegra-spdif-objs := tegra_spdif.o snd-soc-tegra-utils-objs += tegra_asoc_utils.o obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-utils.o obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-das.o obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-pcm.o obj-$(CONFIG_SND_SOC_TEGRA_I2S) += snd-soc-tegra-i2s.o +obj-$(CONFIG_SND_SOC_TEGRA_SPDIF) += snd-soc-tegra-spdif.o # Tegra machine Support snd-soc-tegra-wm8903-objs := tegra_wm8903.o diff --git a/sound/soc/tegra/tegra_spdif.c b/sound/soc/tegra/tegra_spdif.c new file mode 100644 index 0000000..abe606b --- /dev/null +++ b/sound/soc/tegra/tegra_spdif.c @@ -0,0 +1,371 @@ +/* + * tegra_spdif.c - Tegra SPDIF driver + * + * Author: Stephen Warren <swarren(a)nvidia.com> + * Copyright (C) 2011 - NVIDIA, Inc. + * + * 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/clk.h> +#include <linux/module.h> +#include <linux/debugfs.h> +#include <linux/device.h> +#include <linux/platform_device.h> +#include <linux/seq_file.h> +#include <linux/slab.h> +#include <linux/io.h> +#include <mach/iomap.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "tegra_spdif.h" + +#define DRV_NAME "tegra-spdif" + +static inline void tegra_spdif_write(struct tegra_spdif *spdif, u32 reg, + u32 val) +{ + __raw_writel(val, spdif->regs + reg); +} + +static inline u32 tegra_spdif_read(struct tegra_spdif *spdif, u32 reg) +{ + return __raw_readl(spdif->regs + reg); +} + +#ifdef CONFIG_DEBUG_FS +static int tegra_spdif_show(struct seq_file *s, void *unused) +{ +#define REG(r) { r, #r } + static const struct { + int offset; + const char *name; + } regs[] = { + REG(TEGRA_SPDIF_CTRL), + REG(TEGRA_SPDIF_STATUS), + REG(TEGRA_SPDIF_STROBE_CTRL), + REG(TEGRA_SPDIF_DATA_FIFO_CSR), + REG(TEGRA_SPDIF_CH_STA_RX_A), + REG(TEGRA_SPDIF_CH_STA_RX_B), + REG(TEGRA_SPDIF_CH_STA_RX_C), + REG(TEGRA_SPDIF_CH_STA_RX_D), + REG(TEGRA_SPDIF_CH_STA_RX_E), + REG(TEGRA_SPDIF_CH_STA_RX_F), + REG(TEGRA_SPDIF_CH_STA_TX_A), + REG(TEGRA_SPDIF_CH_STA_TX_B), + REG(TEGRA_SPDIF_CH_STA_TX_C), + REG(TEGRA_SPDIF_CH_STA_TX_D), + REG(TEGRA_SPDIF_CH_STA_TX_E), + REG(TEGRA_SPDIF_CH_STA_TX_F), + }; +#undef REG + + struct tegra_spdif *spdif = s->private; + int i; + + for (i = 0; i < ARRAY_SIZE(regs); i++) { + u32 val = tegra_spdif_read(spdif, regs[i].offset); + seq_printf(s, "%s = %08x\n", regs[i].name, val); + } + + return 0; +} + +static int tegra_spdif_debug_open(struct inode *inode, struct file *file) +{ + return single_open(file, tegra_spdif_show, inode->i_private); +} + +static const struct file_operations tegra_spdif_debug_fops = { + .open = tegra_spdif_debug_open, + .read = seq_read, + .llseek = seq_lseek, + .release = single_release, +}; + +static void tegra_spdif_debug_add(struct tegra_spdif *spdif) +{ + spdif->debug = debugfs_create_file(DRV_NAME, S_IRUGO, + snd_soc_debugfs_root, spdif, + &tegra_spdif_debug_fops); +} + +static void tegra_spdif_debug_remove(struct tegra_spdif *spdif) +{ + if (spdif->debug) + debugfs_remove(spdif->debug); +} +#else +static inline void tegra_spdif_debug_add(struct tegra_spdif *spdif) +{ +} + +static inline void tegra_spdif_debug_remove(struct tegra_spdif *spdif) +{ +} +#endif + +static int tegra_spdif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct device *dev = substream->pcm->card->dev; + struct tegra_spdif *spdif = snd_soc_dai_get_drvdata(dai); + int ret, srate, spdifclock; + + spdif->reg_ctrl &= ~TEGRA_SPDIF_CTRL_PACK; + spdif->reg_ctrl &= ~TEGRA_SPDIF_CTRL_BIT_MODE_MASK; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + spdif->reg_ctrl |= TEGRA_SPDIF_CTRL_PACK; + spdif->reg_ctrl |= TEGRA_SPDIF_CTRL_BIT_MODE_16BIT; + break; + default: + return -EINVAL; + } + + srate = params_rate(params); + switch (params_rate(params)) { + case 32000: + spdifclock = 4096000; + break; + case 44100: + spdifclock = 5644800; + break; + case 48000: + spdifclock = 6144000; + break; + case 88200: + spdifclock = 11289600; + break; + case 96000: + spdifclock = 12288000; + break; + case 176400: + spdifclock = 22579200; + break; + case 192000: + spdifclock = 24576000; + break; + default: + return -EINVAL; + } + + ret = clk_set_rate(spdif->clk_spdif_out, spdifclock); + if (ret) { + dev_err(dev, "Can't set SPDIF clock rate: %d\n", ret); + return ret; + } + + return 0; +} + +static void tegra_spdif_start_playback(struct tegra_spdif *spdif) +{ + spdif->reg_ctrl |= TEGRA_SPDIF_CTRL_TX_EN; + tegra_spdif_write(spdif, TEGRA_SPDIF_CTRL, spdif->reg_ctrl); +} + +static void tegra_spdif_stop_playback(struct tegra_spdif *spdif) +{ + spdif->reg_ctrl &= ~TEGRA_SPDIF_CTRL_TX_EN; + tegra_spdif_write(spdif, TEGRA_SPDIF_CTRL, spdif->reg_ctrl); +} + +static int tegra_spdif_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct tegra_spdif *spdif = snd_soc_dai_get_drvdata(dai); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + if (!spdif->clk_refs) + clk_enable(spdif->clk_spdif_out); + spdif->clk_refs++; + tegra_spdif_start_playback(spdif); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + tegra_spdif_stop_playback(spdif); + spdif->clk_refs--; + if (!spdif->clk_refs) + clk_disable(spdif->clk_spdif_out); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int tegra_spdif_probe(struct snd_soc_dai *dai) +{ + struct tegra_spdif *spdif = snd_soc_dai_get_drvdata(dai); + + dai->capture_dma_data = NULL; + dai->playback_dma_data = &spdif->playback_dma_data; + + return 0; +} + +static struct snd_soc_dai_ops tegra_spdif_dai_ops = { + .hw_params = tegra_spdif_hw_params, + .trigger = tegra_spdif_trigger, +}; + +struct snd_soc_dai_driver tegra_spdif_dai = { + .name = DRV_NAME, + .probe = tegra_spdif_probe, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &tegra_spdif_dai_ops, +}; + +static __devinit int tegra_spdif_platform_probe(struct platform_device *pdev) +{ + struct tegra_spdif *spdif; + struct resource *mem, *memregion, *dmareq; + int ret; + + spdif = kzalloc(sizeof(struct tegra_spdif), GFP_KERNEL); + if (!spdif) { + dev_err(&pdev->dev, "Can't allocate tegra_spdif\n"); + ret = -ENOMEM; + goto exit; + } + dev_set_drvdata(&pdev->dev, spdif); + + spdif->clk_spdif_out = clk_get(&pdev->dev, "spdif_out"); + if (IS_ERR(spdif->clk_spdif_out)) { + pr_err("Can't retrieve spdif clock\n"); + ret = PTR_ERR(spdif->clk_spdif_out); + goto err_free; + } + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem) { + dev_err(&pdev->dev, "No memory resource\n"); + ret = -ENODEV; + goto err_clk_put; + } + + dmareq = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmareq) { + dev_err(&pdev->dev, "No DMA resource\n"); + ret = -ENODEV; + goto err_clk_put; + } + + memregion = request_mem_region(mem->start, resource_size(mem), + DRV_NAME); + if (!memregion) { + dev_err(&pdev->dev, "Memory region already claimed\n"); + ret = -EBUSY; + goto err_clk_put; + } + + spdif->regs = ioremap(mem->start, resource_size(mem)); + if (!spdif->regs) { + dev_err(&pdev->dev, "ioremap failed\n"); + ret = -ENOMEM; + goto err_release; + } + + spdif->playback_dma_data.addr = mem->start + TEGRA_SPDIF_DATA_OUT; + spdif->playback_dma_data.wrap = 4; + spdif->playback_dma_data.width = 32; + spdif->playback_dma_data.req_sel = dmareq->start; + + ret = snd_soc_register_dai(&pdev->dev, &tegra_spdif_dai); + if (ret) { + dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); + ret = -ENOMEM; + goto err_unmap; + } + + tegra_spdif_debug_add(spdif); + + return 0; + +err_unmap: + iounmap(spdif->regs); +err_release: + release_mem_region(mem->start, resource_size(mem)); +err_clk_put: + clk_put(spdif->clk_spdif_out); +err_free: + kfree(spdif); +exit: + return ret; +} + +static int __devexit tegra_spdif_platform_remove(struct platform_device *pdev) +{ + struct tegra_spdif *spdif = dev_get_drvdata(&pdev->dev); + struct resource *res; + + snd_soc_unregister_dai(&pdev->dev); + + tegra_spdif_debug_remove(spdif); + + iounmap(spdif->regs); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + release_mem_region(res->start, resource_size(res)); + + clk_put(spdif->clk_spdif_out); + + kfree(spdif); + + return 0; +} + +static struct platform_driver tegra_spdif_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + }, + .probe = tegra_spdif_platform_probe, + .remove = __devexit_p(tegra_spdif_platform_remove), +}; + +static int __init snd_tegra_spdif_init(void) +{ + return platform_driver_register(&tegra_spdif_driver); +} +module_init(snd_tegra_spdif_init); + +static void __exit snd_tegra_spdif_exit(void) +{ + platform_driver_unregister(&tegra_spdif_driver); +} +module_exit(snd_tegra_spdif_exit); + +MODULE_AUTHOR("Stephen Warren <swarren(a)nvidia.com>"); +MODULE_DESCRIPTION("Tegra SPDIF ASoC driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/tegra/tegra_spdif.h b/sound/soc/tegra/tegra_spdif.h new file mode 100644 index 0000000..2e03db4 --- /dev/null +++ b/sound/soc/tegra/tegra_spdif.h @@ -0,0 +1,473 @@ +/* + * tegra_spdif.h - Definitions for Tegra SPDIF driver + * + * Author: Stephen Warren <swarren(a)nvidia.com> + * Copyright (C) 2011 - NVIDIA, Inc. + * + * Based on code copyright/by: + * Copyright (c) 2008-2009, NVIDIA Corporation + * + * 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 + * + */ + +#ifndef __TEGRA_SPDIF_H__ +#define __TEGRA_SPDIF_H__ + +#include "tegra_pcm.h" + +/* Offsets from TEGRA_SPDIF_BASE */ + +#define TEGRA_SPDIF_CTRL 0x0 +#define TEGRA_SPDIF_STATUS 0x4 +#define TEGRA_SPDIF_STROBE_CTRL 0x8 +#define TEGRA_SPDIF_DATA_FIFO_CSR 0x0C +#define TEGRA_SPDIF_DATA_OUT 0x40 +#define TEGRA_SPDIF_DATA_IN 0x80 +#define TEGRA_SPDIF_CH_STA_RX_A 0x100 +#define TEGRA_SPDIF_CH_STA_RX_B 0x104 +#define TEGRA_SPDIF_CH_STA_RX_C 0x108 +#define TEGRA_SPDIF_CH_STA_RX_D 0x10C +#define TEGRA_SPDIF_CH_STA_RX_E 0x110 +#define TEGRA_SPDIF_CH_STA_RX_F 0x114 +#define TEGRA_SPDIF_CH_STA_TX_A 0x140 +#define TEGRA_SPDIF_CH_STA_TX_B 0x144 +#define TEGRA_SPDIF_CH_STA_TX_C 0x148 +#define TEGRA_SPDIF_CH_STA_TX_D 0x14C +#define TEGRA_SPDIF_CH_STA_TX_E 0x150 +#define TEGRA_SPDIF_CH_STA_TX_F 0x154 +#define TEGRA_SPDIF_USR_STA_RX_A 0x180 +#define TEGRA_SPDIF_USR_DAT_TX_A 0x1C0 + +/* Fields in TEGRA_SPDIF_CTRL */ + +/* Start capturing from 0=right, 1=left channel */ +#define TEGRA_SPDIF_CTRL_CAP_LC (1 << 30) + +/* SPDIF receiver(RX) enable */ +#define TEGRA_SPDIF_CTRL_RX_EN (1 << 29) + +/* SPDIF Transmitter(TX) enable */ +#define TEGRA_SPDIF_CTRL_TX_EN (1 << 28) + +/* Transmit Channel status */ +#define TEGRA_SPDIF_CTRL_TC_EN (1 << 27) + +/* Transmit user Data */ +#define TEGRA_SPDIF_CTRL_TU_EN (1 << 26) + +/* Interrupt on transmit error */ +#define TEGRA_SPDIF_CTRL_IE_TXE (1 << 25) + +/* Interrupt on receive error */ +#define TEGRA_SPDIF_CTRL_IE_RXE (1 << 24) + +/* Interrupt on invalid preamble */ +#define TEGRA_SPDIF_CTRL_IE_P (1 << 23) + +/* Interrupt on "B" preamble */ +#define TEGRA_SPDIF_CTRL_IE_B (1 << 22) + +/* Interrupt when block of channel status received */ +#define TEGRA_SPDIF_CTRL_IE_C (1 << 21) + +/* Interrupt when a valid information unit (IU) is received */ +#define TEGRA_SPDIF_CTRL_IE_U (1 << 20) + +/* Interrupt when RX user FIFO attention level is reached */ +#define TEGRA_SPDIF_CTRL_QE_RU (1 << 19) + +/* Interrupt when TX user FIFO attention level is reached */ +#define TEGRA_SPDIF_CTRL_QE_TU (1 << 18) + +/* Interrupt when RX data FIFO attention level is reached */ +#define TEGRA_SPDIF_CTRL_QE_RX (1 << 17) + +/* Interrupt when TX data FIFO attention level is reached */ +#define TEGRA_SPDIF_CTRL_QE_TX (1 << 16) + +/* Loopback test mode enable */ +#define TEGRA_SPDIF_CTRL_LBK_EN (1 << 15) + +/* + * Pack data mode: + * 0 = Single data (16 bit needs to be padded to match the + * interface data bit size). + * 1 = Packeted left/right channel data into a single word. + */ +#define TEGRA_SPDIF_CTRL_PACK (1 << 14) + +/* + * 00 = 16bit data + * 01 = 20bit data + * 10 = 24bit data + * 11 = raw data + */ +#define TEGRA_SPDIF_BIT_MODE_16BIT 0 +#define TEGRA_SPDIF_BIT_MODE_20BIT 1 +#define TEGRA_SPDIF_BIT_MODE_24BIT 2 +#define TEGRA_SPDIF_BIT_MODE_RAW 3 + +#define TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT 12 +#define TEGRA_SPDIF_CTRL_BIT_MODE_MASK (3 << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT) +#define TEGRA_SPDIF_CTRL_BIT_MODE_16BIT (TEGRA_SPDIF_BIT_MODE_16BIT << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT) +#define TEGRA_SPDIF_CTRL_BIT_MODE_20BIT (TEGRA_SPDIF_BIT_MODE_20BIT << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT) +#define TEGRA_SPDIF_CTRL_BIT_MODE_24BIT (TEGRA_SPDIF_BIT_MODE_24BIT << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT) +#define TEGRA_SPDIF_CTRL_BIT_MODE_RAW (TEGRA_SPDIF_BIT_MODE_RAW << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT) + +/* Fields in TEGRA_SPDIF_STATUS */ + +/* + * Note: IS_P, IS_B, IS_C, and IS_U are sticky bits. Software must + * write a 1 to the corresponding bit location to clear the status. + */ + +/* + * Receiver(RX) shifter is busy receiving data. + * This bit is asserted when the receiver first locked onto the + * preamble of the data stream after RX_EN is asserted. This bit is + * deasserted when either, + * (a) the end of a frame is reached after RX_EN is deeasserted, or + * (b) the SPDIF data stream becomes inactive. + */ +#define TEGRA_SPDIF_STATUS_RX_BSY (1 << 29) + +/* + * Transmitter(TX) shifter is busy transmitting data. + * This bit is asserted when TX_EN is asserted. + * This bit is deasserted when the end of a frame is reached after + * TX_EN is deasserted. + */ +#define TEGRA_SPDIF_STATUS_TX_BSY (1 << 28) + +/* + * TX is busy shifting out channel status. + * This bit is asserted when both TX_EN and TC_EN are asserted and + * data from CH_STA_TX_A register is loaded into the internal shifter. + * This bit is deasserted when either, + * (a) the end of a frame is reached after TX_EN is deasserted, or + * (b) CH_STA_TX_F register is loaded into the internal shifter. + */ +#define TEGRA_SPDIF_STATUS_TC_BSY (1 << 27) + +/* + * TX User data FIFO busy. + * This bit is asserted when TX_EN and TXU_EN are asserted and + * there's data in the TX user FIFO. This bit is deassert when either, + * (a) the end of a frame is reached after TX_EN is deasserted, or + * (b) there's no data left in the TX user FIFO. + */ +#define TEGRA_SPDIF_STATUS_TU_BSY (1 << 26) + +/* TX FIFO Underrun error status */ +#define TEGRA_SPDIF_STATUS_TX_ERR (1 << 25) + +/* RX FIFO Overrun error status */ +#define TEGRA_SPDIF_STATUS_RX_ERR (1 << 24) + +/* Preamble status: 0=Preamble OK, 1=bad/missing preamble */ +#define TEGRA_SPDIF_STATUS_IS_P (1 << 23) + +/* B-preamble detection status: 0=not detected, 1=B-preamble detected */ +#define TEGRA_SPDIF_STATUS_IS_B (1 << 22) + +/* + * RX channel block data receive status: + * 0=entire block not recieved yet. + * 1=received entire block of channel status, + */ +#define TEGRA_SPDIF_STATUS_IS_C (1 << 21) + +/* RX User Data Valid flag: 1=valid IU detected, 0 = no IU detected. */ +#define TEGRA_SPDIF_STATUS_IS_U (1 << 20) + +/* + * RX User FIFO Status: + * 1=attention level reached, 0=attention level not reached. + */ +#define TEGRA_SPDIF_STATUS_QS_RU (1 << 19) + +/* + * TX User FIFO Status: + * 1=attention level reached, 0=attention level not reached. + */ +#define TEGRA_SPDIF_STATUS_QS_TU (1 << 18) + +/* + * RX Data FIFO Status: + * 1=attention level reached, 0=attention level not reached. + */ +#define TEGRA_SPDIF_STATUS_QS_RX (1 << 17) + +/* + * TX Data FIFO Status: + * 1=attention level reached, 0=attention level not reached. + */ +#define TEGRA_SPDIF_STATUS_QS_TX (1 << 16) + +/* Fields in TEGRA_SPDIF_STROBE_CTRL */ + +/* + * Indicates the approximate number of detected SPDIFIN clocks within a + * bi-phase period. + */ +#define TEGRA_SPDIF_STROBE_CTRL_PERIOD_SHIFT 16 +#define TEGRA_SPDIF_STROBE_CTRL_PERIOD_MASK (0xff << TEGRA_SPDIF_STROBE_CTRL_PERIOD_SHIFT) + +/* Data strobe mode: 0=Auto-locked 1=Manual locked */ +#define TEGRA_SPDIF_STROBE_CTRL_STROBE (1 << 15) + +/* + * Manual data strobe time within the bi-phase clock period (in terms of + * the number of over-sampling clocks). + */ +#define TEGRA_SPDIF_STROBE_CTRL_DATA_STROBES_SHIFT 8 +#define TEGRA_SPDIF_STROBE_CTRL_DATA_STROBES_MASK (0x1f << TEGRA_SPDIF_STROBE_CTRL_DATA_STROBES_SHIFT) + +/* + * Manual SPDIFIN bi-phase clock period (in terms of the number of + * over-sampling clocks). + */ +#define TEGRA_SPDIF_STROBE_CTRL_CLOCK_PERIOD_SHIFT 0 +#define TEGRA_SPDIF_STROBE_CTRL_CLOCK_PERIOD_MASK (0x3f << TEGRA_SPDIF_STROBE_CTRL_CLOCK_PERIOD_SHIFT) + +/* Fields in SPDIF_DATA_FIFO_CSR */ + +/* Clear Receiver User FIFO (RX USR.FIFO) */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_CLR (1 << 31) + +#define TEGRA_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT 0 +#define TEGRA_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS 1 +#define TEGRA_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS 2 +#define TEGRA_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS 3 + +/* RU FIFO attention level */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT 29 +#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_MASK \ + (0x3 << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU1_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU2_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU3_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU4_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) + +/* Number of RX USR.FIFO levels with valid data. */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_SHIFT 24 +#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_MASK (0x1f << TEGRA_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_SHIFT) + +/* Clear Transmitter User FIFO (TX USR.FIFO) */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_CLR (1 << 23) + +/* TU FIFO attention level */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT 21 +#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_MASK \ + (0x3 << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU1_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU2_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU3_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU4_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) + +/* Number of TX USR.FIFO levels that could be filled. */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_SHIFT 16 +#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_MASK (0x1f << SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_SHIFT) + +/* Clear Receiver Data FIFO (RX DATA.FIFO) */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_CLR (1 << 15) + +#define TEGRA_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT 0 +#define TEGRA_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS 1 +#define TEGRA_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS 2 +#define TEGRA_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS 3 + +/* RU FIFO attention level */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT 13 +#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_MASK \ + (0x3 << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU1_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU4_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU8_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU12_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) + +/* Number of RX DATA.FIFO levels with valid data. */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_SHIFT 8 +#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_MASK (0x1f << TEGRA_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_SHIFT) + +/* Clear Transmitter Data FIFO (TX DATA.FIFO) */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_CLR (1 << 7) + +/* TU FIFO attention level */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT 5 +#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_MASK \ + (0x3 << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU1_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU4_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU8_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU12_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) + +/* Number of TX DATA.FIFO levels that could be filled. */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_SHIFT 0 +#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_MASK (0x1f << SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_SHIFT) + +/* Fields in TEGRA_SPDIF_DATA_OUT */ + +/* + * This register has 5 different formats: + * 16-bit (BIT_MODE=00, PACK=0) + * 20-bit (BIT_MODE=01, PACK=0) + * 24-bit (BIT_MODE=10, PACK=0) + * raw (BIT_MODE=11, PACK=0) + * 16-bit packed (BIT_MODE=00, PACK=1) + */ + +#define TEGRA_SPDIF_DATA_OUT_DATA_16_SHIFT 0 +#define TEGRA_SPDIF_DATA_OUT_DATA_16_MASK (0xffff << TEGRA_SPDIF_DATA_OUT_DATA_16_SHIFT) + +#define TEGRA_SPDIF_DATA_OUT_DATA_20_SHIFT 0 +#define TEGRA_SPDIF_DATA_OUT_DATA_20_MASK (0xfffff << TEGRA_SPDIF_DATA_OUT_DATA_20_SHIFT) + +#define TEGRA_SPDIF_DATA_OUT_DATA_24_SHIFT 0 +#define TEGRA_SPDIF_DATA_OUT_DATA_24_MASK (0xffffff << TEGRA_SPDIF_DATA_OUT_DATA_24_SHIFT) + +#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_P (1 << 31) +#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_C (1 << 30) +#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_U (1 << 29) +#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_V (1 << 28) + +#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_DATA_SHIFT 8 +#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_DATA_MASK (0xfffff << TEGRA_SPDIF_DATA_OUT_DATA_RAW_DATA_SHIFT) + +#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_AUX_SHIFT 4 +#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_AUX_MASK (0xf << TEGRA_SPDIF_DATA_OUT_DATA_RAW_AUX_SHIFT) + +#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_SHIFT 0 +#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_MASK (0xf << TEGRA_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_SHIFT) + +#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_SHIFT 16 +#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_MASK (0xffff << TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_SHIFT) + +#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_SHIFT 0 +#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_MASK (0xffff << TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_SHIFT) + +/* Fields in TEGRA_SPDIF_DATA_IN */ + +/* + * This register has 5 different formats: + * 16-bit (BIT_MODE=00, PACK=0) + * 20-bit (BIT_MODE=01, PACK=0) + * 24-bit (BIT_MODE=10, PACK=0) + * raw (BIT_MODE=11, PACK=0) + * 16-bit packed (BIT_MODE=00, PACK=1) + * + * Bits 31:24 are common to all modes except 16-bit packed + */ + +#define TEGRA_SPDIF_DATA_IN_DATA_P (1 << 31) +#define TEGRA_SPDIF_DATA_IN_DATA_C (1 << 30) +#define TEGRA_SPDIF_DATA_IN_DATA_U (1 << 29) +#define TEGRA_SPDIF_DATA_IN_DATA_V (1 << 28) + +#define TEGRA_SPDIF_DATA_IN_DATA_PREAMBLE_SHIFT 24 +#define TEGRA_SPDIF_DATA_IN_DATA_PREAMBLE_MASK (0xf << TEGRA_SPDIF_DATA_IN_DATA_PREAMBLE_SHIFT) + +#define TEGRA_SPDIF_DATA_IN_DATA_16_SHIFT 0 +#define TEGRA_SPDIF_DATA_IN_DATA_16_MASK (0xffff << TEGRA_SPDIF_DATA_IN_DATA_16_SHIFT) + +#define TEGRA_SPDIF_DATA_IN_DATA_20_SHIFT 0 +#define TEGRA_SPDIF_DATA_IN_DATA_20_MASK (0xfffff << TEGRA_SPDIF_DATA_IN_DATA_20_SHIFT) + +#define TEGRA_SPDIF_DATA_IN_DATA_24_SHIFT 0 +#define TEGRA_SPDIF_DATA_IN_DATA_24_MASK (0xffffff << TEGRA_SPDIF_DATA_IN_DATA_24_SHIFT) + +#define TEGRA_SPDIF_DATA_IN_DATA_RAW_DATA_SHIFT 8 +#define TEGRA_SPDIF_DATA_IN_DATA_RAW_DATA_MASK (0xfffff << TEGRA_SPDIF_DATA_IN_DATA_RAW_DATA_SHIFT) + +#define TEGRA_SPDIF_DATA_IN_DATA_RAW_AUX_SHIFT 4 +#define TEGRA_SPDIF_DATA_IN_DATA_RAW_AUX_MASK (0xf << TEGRA_SPDIF_DATA_IN_DATA_RAW_AUX_SHIFT) + +#define TEGRA_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_SHIFT 0 +#define TEGRA_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_MASK (0xf << TEGRA_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_SHIFT) + +#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_SHIFT 16 +#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_MASK (0xffff << TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_SHIFT) + +#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_SHIFT 0 +#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_MASK (0xffff << TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_SHIFT) + +/* Fields in TEGRA_SPDIF_CH_STA_RX_A */ +/* Fields in TEGRA_SPDIF_CH_STA_RX_B */ +/* Fields in TEGRA_SPDIF_CH_STA_RX_C */ +/* Fields in TEGRA_SPDIF_CH_STA_RX_D */ +/* Fields in TEGRA_SPDIF_CH_STA_RX_E */ +/* Fields in TEGRA_SPDIF_CH_STA_RX_F */ + +/* + * The 6-word receive channel data page buffer holds a block (192 frames) of + * channel status information. The order of receive is from LSB to MSB + * bit, and from CH_STA_RX_A to CH_STA_RX_F then back to CH_STA_RX_A. + */ + +/* Fields in TEGRA_SPDIF_CH_STA_TX_A */ +/* Fields in TEGRA_SPDIF_CH_STA_TX_B */ +/* Fields in TEGRA_SPDIF_CH_STA_TX_C */ +/* Fields in TEGRA_SPDIF_CH_STA_TX_D */ +/* Fields in TEGRA_SPDIF_CH_STA_TX_E */ +/* Fields in TEGRA_SPDIF_CH_STA_TX_F */ + +/* + * The 6-word transmit channel data page buffer holds a block (192 frames) of + * channel status information. The order of transmission is from LSB to MSB + * bit, and from CH_STA_TX_A to CH_STA_TX_F then back to CH_STA_TX_A. + */ + +/* Fields in TEGRA_SPDIF_USR_STA_RX_A */ + +/* + * This 4-word deep FIFO receives user FIFO field information. The order of + * receive is from LSB to MSB bit. + */ + +/* Fields in TEGRA_SPDIF_USR_DAT_TX_A */ + +/* + * This 4-word deep FIFO transmits user FIFO field information. The order of + * transmission is from LSB to MSB bit. + */ + +struct tegra_spdif { + struct clk *clk_spdif_out; + int clk_refs; + struct tegra_pcm_dma_params capture_dma_data; + struct tegra_pcm_dma_params playback_dma_data; + void __iomem *regs; + struct dentry *debug; + u32 reg_ctrl; +}; + +#endif -- 1.7.0.4

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[alsa-devel] [PATCH] ASoC: core - Add platform IO tracing
by Liam Girdwood 05 Jul '11

05 Jul '11

Trace platform IO just like CODEC IO. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/trace/events/asoc.h | 45 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/soc-core.c | 2 + 2 files changed, 47 insertions(+), 0 deletions(-) diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h index ae973d2..603f5a0 100644 --- a/include/trace/events/asoc.h +++ b/include/trace/events/asoc.h @@ -9,6 +9,7 @@ struct snd_soc_jack; struct snd_soc_codec; +struct snd_soc_platform; struct snd_soc_card; struct snd_soc_dapm_widget; @@ -59,6 +60,50 @@ DEFINE_EVENT(snd_soc_reg, snd_soc_reg_read, ); +DECLARE_EVENT_CLASS(snd_soc_preg, + + TP_PROTO(struct snd_soc_platform *platform, unsigned int reg, + unsigned int val), + + TP_ARGS(platform, reg, val), + + TP_STRUCT__entry( + __string( name, platform->name ) + __field( int, id ) + __field( unsigned int, reg ) + __field( unsigned int, val ) + ), + + TP_fast_assign( + __assign_str(name, platform->name); + __entry->id = platform->id; + __entry->reg = reg; + __entry->val = val; + ), + + TP_printk("platform=%s.%d reg=%x val=%x", __get_str(name), + (int)__entry->id, (unsigned int)__entry->reg, + (unsigned int)__entry->val) +); + +DEFINE_EVENT(snd_soc_preg, snd_soc_preg_write, + + TP_PROTO(struct snd_soc_platform *platform, unsigned int reg, + unsigned int val), + + TP_ARGS(platform, reg, val) + +); + +DEFINE_EVENT(snd_soc_preg, snd_soc_preg_read, + + TP_PROTO(struct snd_soc_platform *platform, unsigned int reg, + unsigned int val), + + TP_ARGS(platform, reg, val) + +); + DECLARE_EVENT_CLASS(snd_soc_card, TP_PROTO(struct snd_soc_card *card, int val), diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d08abf4..06c9b14 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1652,6 +1652,7 @@ int snd_soc_platform_read(struct snd_soc_platform *platform, ret = platform->driver->read(platform, reg); dev_dbg(platform->dev, "read %x => %x\n", reg, ret); + trace_snd_soc_preg_read(platform, reg, ret); return ret; } @@ -1666,6 +1667,7 @@ int snd_soc_platform_write(struct snd_soc_platform *platform, } dev_dbg(platform->dev, "write %x = %x\n", reg, val); + trace_snd_soc_preg_write(platform, reg, val); return platform->driver->write(platform, reg, val); } EXPORT_SYMBOL_GPL(snd_soc_platform_write); -- 1.7.4.1

2 4

[alsa-devel] [PATCH v2 0/5] MFD/input/ASoC: twl6040: irq registration changes
by Peter Ujfalusi 05 Jul '11

05 Jul '11

Hello, Changes since v1: - IRQ number is passed with resource to the slave devices Intro mail froim v1: As Samuel, and Felipe requested for the twl6040 MFD series: http://marc.info/?l=linux-kernel&m=130978512225430&w=2 http://marc.info/?l=linux-omap&m=130978403224218&w=2 This is the update to remove the obfuscation around the threaded irq request in the twl6040 driver and it's slaves. With this change the slaves uses directly the threaded irq request, and the core driver passes the irq number for the slaves to use (configured runtime, so machine drivers do not need to be updated). Regards, Peter PS: for the remaining comments from Samuel I have changed the series, since the requested change was small. I'm not intended to send the series again, unless it is requested (again the change from v6 is minimal - few lines has been removed/replaced). --- Peter Ujfalusi (5): MFD: twl6040: Use resource to provide irq number for slaves input: twl6040-vibra: Do not use wrapper for irq request ASoC: twl6040: Do not use wrapper for irq request MFD: twl6040: Demand valid interrupt configuration MFD: twl6040: Remove wrapper for threaded irq request drivers/input/misc/twl6040-vibra.c | 17 ++++++--- drivers/mfd/twl6040-core.c | 69 ++++++++++++++++++++++++----------- drivers/mfd/twl6040-irq.c | 16 +-------- include/linux/mfd/twl6040.h | 22 ----------- sound/soc/codecs/twl6040.c | 19 +++++++--- 5 files changed, 74 insertions(+), 69 deletions(-) -- 1.7.6

2 6

Re: [alsa-devel] [PATCH v2 2/5] input: twl6040-vibra: Do not use wrapper for irq request
by Péter Ujfalusi 05 Jul '11

05 Jul '11

On Tuesday 05 July 2011 15:57:46 Balbi, Felipe wrote: > > + info->irq = platform_get_irq(pdev, 0); > > + if (!info->irq) { > > this is wrong. platform_get_irq() will return -ENXIO if it can't find > the IRQ you requested: > > 55 /** > 56 * platform_get_irq - get an IRQ for a device > 57 * @dev: platform device > 58 * @num: IRQ number index > 59 */ > 60 int platform_get_irq(struct platform_device *dev, unsigned int num) > 61 { > 62 struct resource *r = platform_get_resource(dev, IORESOURCE_IRQ, > num); 63 > 64 return r ? r->start : -ENXIO; > 65 } > 66 EXPORT_SYMBOL_GPL(platform_get_irq); Yes, I have not checked that.. Will replace the check with (info->irq< 0) checks in both slaves. Thanks for noticing it! I'll wait a bit for other comments, and will resend. -- Péter

1 0

Re: [alsa-devel] [PATCH 1/5] MFD: twl6040: Add irq variable for slave devices
by Péter Ujfalusi 05 Jul '11

05 Jul '11

On Tuesday 05 July 2011 13:22:46 Balbi, Felipe wrote: > Hi, > > On Tue, Jul 05, 2011 at 12:14:22PM +0300, Peter Ujfalusi wrote: > > Add new variable to pass the irq number for the slaves > > devices of twl6040. > > The irq number configuration is done in the twl6040-core > > at probe time, so machine drivers do not need to be > > modified. > > > > Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> > > --- > > > > drivers/mfd/twl6040-core.c | 2 ++ > > include/linux/i2c/twl.h | 2 ++ > > 2 files changed, 4 insertions(+), 0 deletions(-) > > > > diff --git a/drivers/mfd/twl6040-core.c b/drivers/mfd/twl6040-core.c > > index cfaedb5..8607793 100644 > > --- a/drivers/mfd/twl6040-core.c > > +++ b/drivers/mfd/twl6040-core.c > > @@ -501,6 +501,7 @@ static int __devinit twl6040_probe(struct > > platform_device *pdev) > > > > if (pdata->codec) { > > > > cell = &twl6040->cells[children]; > > cell->name = "twl6040-codec"; > > > > + pdata->codec->plug_irq = twl6040->irq_base + TWL6040_IRQ_PLUG; > > are the slave devices platform_devices ?? If so, you could use struct > resource to pass IRQs as that's the "approved" way to do so. Then, on > child driver, you could either rely on a particular order of the > resources or assign them names and use platform_get_irq_byname() to > fetch the correct one (if you have more than one, that is). You mean something like this: static struct resource twl6040_codec_rsrc[] = { { .flags = IORESOURCE_IRQ, }, }; ... @@ -501,6 +501,7 @@ static int __devinit twl6040_probe(struct platform_device *pdev) if (pdata->codec) { cell = &twl6040->cells[children]; cell->name = "twl6040-codec"; + twl6040_codec_rsrc[0].start = twl6040->irq_base + TWL6040_IRQ_PLUG; + twl6040_codec_rsrc[0].end = twl6040->irq_base + TWL6040_IRQ_PLUG; + cell->resource = twl6040_codec_rsrc; + cell->num_resources = ARRAY_SIZE(twl6040_codec_rsrc); cell->platform_data = pdata->codec; cell->pdata_size = sizeof(*pdata->codec); children++; -- Péter

1 0

[alsa-devel] Recording doesn't end gracefully
by Santosh Sivaraj 05 Jul '11

05 Jul '11

Hi all, We were working on 1.0.23 version of the ALSA driver. In the following scenario we are getting some issues with recording. * Play some music in the sound player in the background * Open recorder and start recording * Since playback and recording can't happen simultaneously, we explicitly reset the codec device and configure it for recording. * In this case, after some time the codec driver is powered off by the soc-core and only silence is recorded. All this were done from the Android UI. When the same is repeated from the command line with aplay and arecord we see that aplay is stopped and recording works fine. When we debugged the problem, we found that the following diff seems to get rid of the above problem: diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c984996..247f4ee 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -773,7 +773,8 @@ int snd_soc_pcm_close(struct snd_pcm_substream *substream) /* Muting the DAC suppresses artifacts caused during digital * shutdown, for example from stopping clocks. */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) || + (substream->stream == SNDRV_PCM_STREAM_CAPTURE)) snd_soc_dai_digital_mute(codec_dai, 1); if (cpu_dai->driver->ops->shutdown) @@ -869,7 +870,8 @@ int snd_soc_pcm_prepare(struct snd_pcm_substream *substream) } /* cancel any delayed stream shutdown that is pending */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + if (((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) || + (substream->stream == SNDRV_PCM_STREAM_CAPTURE)) && codec_dai->pop_wait) { codec_dai->pop_wait = 0; cancel_delayed_work(&rtd->delayed_work); I am not aware of what consequences/side effects these changes can cause, please advice, whether it is a problem elsewhere -- http://fossix.org, http://dot.slashb.in Things do not change; we change -- Henry David

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Re: [alsa-devel] [PATCH v6 08/18] mfd: twl6040: Add initial support
by Péter Ujfalusi 05 Jul '11

05 Jul '11

Hi Samuel, On Monday 04 July 2011 19:39:35 Samuel Ortiz wrote: > That is fine with me, yes. That's why I ACK the MFD patches for Tony to take > them. I have changed the series for Tony in my branch for the following comments: wl6040_is_powered, twl6040_get_rev removal, since these are really small changes, if you want to see it before Tony pulls it, I can send the selected patches (few lines has been changed). > Thanks in advance for the follow up patches. For the twl6040_*_irq changes, I sent the series, since I ended up with a bigger change than I would have expected. Regards, Péter

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[alsa-devel] [PATCH 0/5] MFD/input/ASoC: twl6040: irq registration changes
by Peter Ujfalusi 05 Jul '11

05 Jul '11

Hello, As Samuel, and Felipe requested for the twl6040 MFD series: http://marc.info/?l=linux-kernel&m=130978512225430&w=2 http://marc.info/?l=linux-omap&m=130978403224218&w=2 This is the update to remove the obfuscation around the threaded irq request in the twl6040 driver and it's slaves. With this change the slaves uses directly the threaded irq request, and the core driver passes the irq number for the slaves to use (configured runtime, so machine drivers do not need to be updated). Regards, Peter PS: for the remaining comments from Samuel I have changed the series, since the requested change was small. I'm not intended to send the series again, unless it is requested (again the change from v6 is minimal - few lines has been removed/replaced). --- Peter Ujfalusi (5): MFD: twl6040: Add irq variable for slave devices input: twl6040-vibra: Do not use wrapper for irq request ASoC: twl6040: Do not use wrapper for irq request MFD: twl6040: Demand valid interrupt configuration MFD: twl6040: Remove wrapper for threaded irq request drivers/input/misc/twl6040-vibra.c | 17 +++++++++---- drivers/mfd/twl6040-core.c | 46 ++++++++++++++++++----------------- drivers/mfd/twl6040-irq.c | 16 +----------- include/linux/i2c/twl.h | 2 + include/linux/mfd/twl6040.h | 22 ----------------- sound/soc/codecs/twl6040.c | 17 +++++++++---- 6 files changed, 51 insertions(+), 69 deletions(-) -- 1.7.6

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[alsa-devel] ASoC updates for 3.1
by Mark Brown 05 Jul '11

05 Jul '11

The following changes since commit 56aa53391059e3730a304da4dd96b7b123b9fb75: Merge branch 'for-3.1' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc (2011-07-05 07:33:23 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.1 Lars-Peter Clausen (2): ASoC: Add ADAV80x codec driver ASoC: Blackfin: Add machine driver for EVAL-ADAV80X boards Liam Girdwood (2): ASoC: core - Make platform probe more like codec probe. ASoC: core - Add platform read and write. Mark Brown (7): ASoC: Fix Blackfin I2S _pointer() implementation return in bounds values ASoC: Correct left/right swap in wm_hubs DC offset correction ASoC: Change WM9081 speaker output enable to _OUT_DRV Merge branch 'for-3.0' into for-3.1 ASoC: Improve error reporting in Speyside WM8962 driver ASoC: Ensure we delay long enough for WM8994 FLL to lock when starting Merge branch 'for-3.0' into for-3.1 Sangbeom Kim (3): ASoC: SAMSUNG: Add WM8994 PCM Machine driver ASoC: SAMSUNG: Move I2S common register definition ASoC: SAMSUNG: Add idma related register definition Stephen Warren (2): ASoC: Tegra: I2S: Ensure clock is enabled when writing regs ASoC: Tegra: I2S: s/clk_get_sys/clk_get/ MAINTAINERS | 1 + include/sound/soc.h | 8 + sound/soc/blackfin/Kconfig | 13 + sound/soc/blackfin/Makefile | 2 + sound/soc/blackfin/bf5xx-i2s-pcm.c | 13 +- sound/soc/blackfin/bfin-eval-adav80x.c | 173 ++++++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/adav80x.c | 951 ++++++++++++++++++++++++++++++++ sound/soc/codecs/adav80x.h | 35 ++ sound/soc/codecs/wm8994.c | 2 + sound/soc/codecs/wm9081.c | 2 +- sound/soc/codecs/wm_hubs.c | 16 +- sound/soc/samsung/Kconfig | 8 + sound/soc/samsung/Makefile | 2 + sound/soc/samsung/i2s-regs.h | 143 +++++ sound/soc/samsung/i2s.c | 104 +---- sound/soc/samsung/smdk_wm8994pcm.c | 176 ++++++ sound/soc/samsung/speyside_wm8962.c | 12 +- sound/soc/soc-core.c | 81 +++- sound/soc/tegra/tegra_i2s.c | 10 +- 21 files changed, 1622 insertions(+), 136 deletions(-) create mode 100644 sound/soc/blackfin/bfin-eval-adav80x.c create mode 100644 sound/soc/codecs/adav80x.c create mode 100644 sound/soc/codecs/adav80x.h create mode 100644 sound/soc/samsung/i2s-regs.h create mode 100644 sound/soc/samsung/smdk_wm8994pcm.c

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[alsa-devel] ASoC updates for 3.0
by Mark Brown 05 Jul '11

05 Jul '11

The following changes since commit 63bc9750167ac765827470aa5aed1e79aeaaad9e: Merge branch 'for-3.0' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into fix/asoc (2011-07-05 07:33:06 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.0 Mark Brown (2): ASoC: Fix Blackfin I2S _pointer() implementation return in bounds values ASoC: Ensure we delay long enough for WM8994 FLL to lock when starting Stephen Warren (1): ASoC: Tegra: I2S: Ensure clock is enabled when writing regs sound/soc/blackfin/bf5xx-i2s-pcm.c | 13 +++++++++++-- sound/soc/codecs/wm8994.c | 2 ++ sound/soc/tegra/tegra_i2s.c | 6 ++++++ 3 files changed, 19 insertions(+), 2 deletions(-)

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[alsa-devel] [GIT PULL] ASoC updates for 3.1
by Liam Girdwood 05 Jul '11

05 Jul '11

The following changes since commit d21685ec258f803d3badae5eae821383a34815a9: Merge branch 'for-2.6.40' into for-2.6.41 (2011-05-30 10:54:18 +0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git for-3.1 Jarkko Nikula (1): ASoC: tlv320aic3x: Add correct hw registers to Line1 cross connect muxes Ricardo Neri (4): ASoC: OMAP: Add CPU DAI driver for HDMI ASoC: OMAP4: Add HDMI Audio machine driver for OMAP4 boards ASoC: OMAP: Update Makefile and Kconfig for HDMI audio ASoC: twl6040 - According to TWL6040 specification, gain start at 6dB and not -6dB. sound/soc/codecs/tlv320aic3x.c | 34 +++++--- sound/soc/codecs/twl6040.c | 4 +- sound/soc/omap/Kconfig | 11 +++ sound/soc/omap/Makefile | 4 + sound/soc/omap/omap-hdmi.c | 158 ++++++++++++++++++++++++++++++++++++++ sound/soc/omap/omap-hdmi.h | 36 +++++++++ sound/soc/omap/omap4-hdmi-card.c | 129 +++++++++++++++++++++++++++++++ 7 files changed, 361 insertions(+), 15 deletions(-) create mode 100644 sound/soc/omap/omap-hdmi.c create mode 100644 sound/soc/omap/omap-hdmi.h create mode 100644 sound/soc/omap/omap4-hdmi-card.c

2 1

[alsa-devel] [GIT PULL] ASoC fixes for 3.0
by Liam Girdwood 05 Jul '11

05 Jul '11

The following changes since commit 4a787a3ff3f419c23ab0a5cef677fa441356b818: Merge branch 'for-2.6.40' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc (2011-05-20 11:25:32 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git for-3.0 Jarkko Nikula (2): ASoC: tlv320aic3x: Don't sync first two registers from register cache ASoC: tlv320aic3x: Do soft reset to codec when going to bias off state Michael Williamson (1): audio: tlv320aic26: fix PLL register configuration sound/soc/codecs/tlv320aic26.c | 14 +++++++++++--- sound/soc/codecs/tlv320aic3x.c | 9 ++++++++- 2 files changed, 19 insertions(+), 4 deletions(-)

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[alsa-devel] [PATCH] ASoC: core - Add platform read and write.
by Liam Girdwood 05 Jul '11

05 Jul '11

In preparation for ASoC Dynamic PCM (AKA DSP) support. Allow platform driver to perform IO. Intended for platform DAPM. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc.h | 8 ++++++++ sound/soc/soc-core.c | 30 ++++++++++++++++++++++++++++++ 2 files changed, 38 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 6424b10..f30f3fe 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -312,6 +312,10 @@ int snd_soc_default_readable_register(struct snd_soc_codec *codec, unsigned int reg); int snd_soc_default_writable_register(struct snd_soc_codec *codec, unsigned int reg); +int snd_soc_platform_read(struct snd_soc_platform *platform, + unsigned int reg); +int snd_soc_platform_write(struct snd_soc_platform *platform, + unsigned int reg, unsigned int val); /* Utility functions to get clock rates from various things */ int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots); @@ -658,6 +662,10 @@ struct snd_soc_platform_driver { /* probe ordering - for components with runtime dependencies */ int probe_order; int remove_order; + + /* platform IO - used for platform DAPM */ + unsigned int (*read)(struct snd_soc_platform *, unsigned int); + int (*write)(struct snd_soc_platform *, unsigned int, unsigned int); }; struct snd_soc_platform { diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d49f0ed..d08abf4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1640,6 +1640,36 @@ int snd_soc_codec_writable_register(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(snd_soc_codec_writable_register); +int snd_soc_platform_read(struct snd_soc_platform *platform, + unsigned int reg) +{ + unsigned int ret; + + if (!platform->driver->read) { + dev_err(platform->dev, "platform has no read back\n"); + return -1; + } + + ret = platform->driver->read(platform, reg); + dev_dbg(platform->dev, "read %x => %x\n", reg, ret); + + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_platform_read); + +int snd_soc_platform_write(struct snd_soc_platform *platform, + unsigned int reg, unsigned int val) +{ + if (!platform->driver->write) { + dev_err(platform->dev, "platform has no write back\n"); + return -1; + } + + dev_dbg(platform->dev, "write %x = %x\n", reg, val); + return platform->driver->write(platform, reg, val); +} +EXPORT_SYMBOL_GPL(snd_soc_platform_write); + /** * snd_soc_new_ac97_codec - initailise AC97 device * @codec: audio codec -- 1.7.4.1

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Re: [alsa-devel] [PATCH] ASoc: mx27vis-aic32x4: Remove unneeded header
by Mark Brown 04 Jul '11

04 Jul '11

On Mon, Jul 04, 2011 at 04:01:24PM -0300, Fabio Estevam wrote: > There is no machine type check on this code, so remove the mach-types header. > > Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> This isn't really useful - if you're going to change something here you should change the code to check for the machine.

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[PATCH] ASoC: Manage WM8731 ACTIVE bit as a supply widget
by Mark Brown 04 Jul '11

04 Jul '11

Now we have supply widgets there's no need to open code the handling of the ACTIVE bit. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8731.c | 29 +++-------------------------- 1 files changed, 3 insertions(+), 26 deletions(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 2dc964b..59d9e59 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -175,6 +175,7 @@ static const struct snd_kcontrol_new wm8731_input_mux_controls = SOC_DAPM_ENUM("Input Select", wm8731_insel_enum); static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = { +SND_SOC_DAPM_SUPPLY("Active",WM8731_ACTIVE, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("OSC", WM8731_PWR, 5, 1, NULL, 0), SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1, &wm8731_output_mixer_controls[0], @@ -204,6 +205,8 @@ static int wm8731_check_osc(struct snd_soc_dapm_widget *source, static const struct snd_soc_dapm_route wm8731_intercon[] = { {"DAC", NULL, "OSC", wm8731_check_osc}, {"ADC", NULL, "OSC", wm8731_check_osc}, + {"DAC", NULL, "ACTIVE"}, + {"ADC", NULL, "ACTIVE"}, /* output mixer */ {"Output Mixer", "Line Bypass Switch", "Line Input"}, @@ -315,29 +318,6 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream, return 0; } -static int wm8731_pcm_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - - /* set active */ - snd_soc_write(codec, WM8731_ACTIVE, 0x0001); - - return 0; -} - -static void wm8731_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - - /* deactivate */ - if (!codec->active) { - udelay(50); - snd_soc_write(codec, WM8731_ACTIVE, 0x0); - } -} - static int wm8731_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; @@ -480,7 +460,6 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, snd_soc_write(codec, WM8731_PWR, reg | 0x0040); break; case SND_SOC_BIAS_OFF: - snd_soc_write(codec, WM8731_ACTIVE, 0x0); snd_soc_write(codec, WM8731_PWR, 0xffff); regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); @@ -496,9 +475,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_FMTBIT_S24_LE) static struct snd_soc_dai_ops wm8731_dai_ops = { - .prepare = wm8731_pcm_prepare, .hw_params = wm8731_hw_params, - .shutdown = wm8731_shutdown, .digital_mute = wm8731_mute, .set_sysclk = wm8731_set_dai_sysclk, .set_fmt = wm8731_set_dai_fmt, -- 1.7.5.4

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[alsa-devel] [PATCH 0/3] ASoC: Tegra: Fixes and SPDIF support
by Stephen Warren 04 Jul '11

04 Jul '11

A couple of small bug-fixes for the existing Tegra SPDIF support, and a new SPDIF driver. I've tested the SPDIF driver on Harmony. However, I'm not submitting the ASoC machine support patch for inclusion in mainline, since it relies on the board files registering some new platform devices, and we can't upstream any board file changes right now. Still, this may be useful for other board vendors to start working with internally. For people interested in testing/using this, I can push the branch I used for testing this on Harmony publicly sometime next week, hopefully. That branch contains the board changes to support this, and the machine driver change. Stephen Warren (3): ASoC: Tegra: I2S: Ensure clock is enabled when writing regs ASoC: Tegra: I2S: s/clk_get_sys/clk_get/ ASoC: Tegra: Implement SPDIF CPU DAI sound/soc/tegra/Kconfig | 9 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_i2s.c | 10 +- sound/soc/tegra/tegra_spdif.c | 371 ++++++++++++++++++++++++++++++++ sound/soc/tegra/tegra_spdif.h | 473 +++++++++++++++++++++++++++++++++++++++++ 5 files changed, 862 insertions(+), 3 deletions(-) create mode 100644 sound/soc/tegra/tegra_spdif.c create mode 100644 sound/soc/tegra/tegra_spdif.h

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Re: [alsa-devel] [PATCH v6 08/18] mfd: twl6040: Add initial support
by Péter Ujfalusi 04 Jul '11

04 Jul '11

Hi Samuel, On Monday 04 July 2011 13:48:44 Samuel Ortiz wrote: > I don't see the value of those 3 inline functions. Removing them would make > the code actually more understandable (especially for the 2nd one). Is it OK for you if I fix these as a follow-up series, so I do not need to rewrite this series, and resend it? -- Péter

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Re: [alsa-devel] [PATCH v6 08/18] mfd: twl6040: Add initial support
by Péter Ujfalusi 04 Jul '11

04 Jul '11

Hi Samuel, On Monday 04 July 2011 13:48:44 Samuel Ortiz wrote: > Hi Peter, > > On Tue, Jun 21, 2011 at 04:39:06PM +0300, Peter Ujfalusi wrote: > > +int twl6040_is_powered(struct twl6040 *twl6040) > > +{ > > + return twl6040->power_count; > > +} > > +EXPORT_SYMBOL(twl6040_is_powered); > > Do we really need to export this one ? No, we don't. At the end we do not need this function at all either. I'm going to remove it. > > +static inline int twl6040_get_rev(struct twl6040 *twl6040) > > +{ > > + return twl6040->rev; > > +} No need for this either, I'll remove it. > > + > > +static inline int twl6040_request_irq(struct twl6040 *twl6040, int irq, > > + irq_handler_t handler, > > + unsigned long irqflags, > > + const char *name, > > + void *data) > > +{ > > + if (!twl6040->irq_base) > > + return -EINVAL; > > + > > + return request_threaded_irq(twl6040->irq_base + irq, NULL, handler, > > + irqflags, name, data); > > +} > > + > > +static inline void twl6040_free_irq(struct twl6040 *twl6040, int irq, > > + void *data) > > +{ > > + if (!twl6040->irq_base) > > + return; > > + > > + free_irq(twl6040->irq_base + irq, data); > > +} > > I don't see the value of those 3 inline functions. Removing them would make > the code actually more understandable (especially for the 2nd one). The twl6040_request_irq is also used by the ASoC codec driver, and the vibra driver, but if you think it is better to replace those as well, I will do it right away. -- Péter

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Re: [alsa-devel] [PATCH v6 08/18] mfd: twl6040: Add initial support
by Péter Ujfalusi 04 Jul '11

04 Jul '11

Hi Felipe, Samuel, On Monday 04 July 2011 14:53:30 Balbi, Felipe wrote: > > + ret = twl6040_request_irq(twl6040, TWL6040_IRQ_READY, > > + twl6040_naudint_handler, 0, > > + "twl6040_irq_ready", twl6040); > > why don't you use the normal request_threaded_irq() ?? This is a bit of > obfuscation IMO. I have left this 'macro' in since it is also used within the twl6040-vibra, and twl6040 ASoC codec driver to request the irq for them. But if you think it is better to replace the twl6040_request_irq with direct request_threaded_irq in those drivers as well, I'll do that. -- Péter

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[alsa-devel] [PATCH] ALSA: hda-intel - add position_fix quirk for Gateway NV53A
by Dmitry Torokhov 04 Jul '11

04 Jul '11

Without this option internal microphone does not seem to be working. $ lspci -vvnn | grep -A10 Audio 00:14.2 Audio device [0403]: ATI Technologies Inc SBx00 Azalia (Intel HDA) [1002:4383] (rev 40) Subsystem: Acer Incorporated [ALI] Device [1025:036e] Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=slow >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64, Cache Line Size: 64 bytes Interrupt: pin ? routed to IRQ 16 Region 0: Memory at d2300000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Kernel driver in use: HDA Intel -- 01:05.1 Audio device [0403]: ATI Technologies Inc RS880 Audio Device [Radeon HD 4200] [1002:970f] Subsystem: Acer Incorporated [ALI] Device [1025:036e] Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin B routed to IRQ 19 Region 0: Memory at d2210000 (32-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 3 Flags: PMEClk- DSI- D1+ D2+ AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Capabilities: [a0] MSI: Enable- Count=1/1 Maskable- 64bit+ Signed-off-by: Dmitry Torokhov <dtor(a)mail.ru> --- sound/pci/hda/hda_intel.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 486f6de..4fbcea6 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2349,6 +2349,7 @@ static int azx_dev_free(struct snd_device *device) static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1025, 0x009f, "Acer Aspire 5110", POS_FIX_LPIB), SND_PCI_QUIRK(0x1025, 0x026f, "Acer Aspire 5538", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1025, 0x036e, "Gateway NV53A", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01f6, "Dell Latitude 131L", POS_FIX_LPIB),

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Re: [alsa-devel] Noise on snd-hda-intel
by Raymond Yau 04 Jul '11

04 Jul '11

2011/6/24 Sune Mølgaard <sune(a)molgaard.org>: Refer to patch eb541337b7a43822fce7d0c9d967ee149b2d9a96 "ALSA: hda - Make converter setups sticky" The patch create an array of 8 elements for converter caches but there are more than 8 converters in ad1988/ad1989 5 analog Audio Output + 3 analog Audio Input + Digital Audio Output + DIgital Input + snd_array_init(&codec->cvt_setups, sizeof(struct hda_cvt_setup), 8); >> >> Is this a clean install of 11.4 or upgrade from previous version of >> ubuntu which use "default:2" and create "Digital Capture Volume" ? > > It is an upgrade. !!Modprobe options (Sound related) !!-------------------------------- snd-usb-audio: index=-2 snd_hda_intel: model=6stack enable=1 index=0 Seem those snd-usb-audio are loaded before snd-hda-intel and that's why your onboard become card 2 instead of card 0 > >> since pulseaudio use "front:2" for playback and capture and this won't >> create "digital capture volume" control > > How to remedy? Are you upgarded from version which use dmix/dsnoop ? or the "Digital Capture Volume" control is created when remove alsa-pulse-plugin There are three capture subdevices since ad1988/ad1989 have 3 [Audio Input] when application using dsnoop which use subdevice 0 by default However for PA server release the capture device after probe so it is possible that other application (e.g. arecord ) can open subdevice 0 PA server will open the next available subdevice https://qa.mandriva.com/show_bug.cgi?id=52175 card 2: Intel [HDA Intel], device 0: AD198x Analog [AD198x Analog] Subdevices: 3/3 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Simple mixer control 'Capture',0 Capabilities: cvolume cswitch penum Capture channels: Front Left - Front Right Limits: Capture 0 - 54 Front Left: Capture 48 [89%] [13.50dB] [on] Front Right: Capture 48 [89%] [13.50dB] [on] Simple mixer control 'Capture',1 Capabilities: cvolume cswitch penum Capture channels: Front Left - Front Right Limits: Capture 0 - 54 Front Left: Capture 0 [0%] [-58.50dB] [off] Front Right: Capture 0 [0%] [-58.50dB] [off] Simple mixer control 'Capture',2 Capabilities: cvolume cswitch penum Capture channels: Front Left - Front Right Limits: Capture 0 - 54 Front Left: Capture 0 [0%] [-58.50dB] [off] Front Right: Capture 0 [0%] [-58.50dB] [off]

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[alsa-devel] [PATCH] ASoC: core - Make platform probe more like codec probe.
by Liam Girdwood 02 Jul '11

02 Jul '11

In preparation for ASoC dynamic PCM support (AKA ASoC DSP) Platform will also support DAPM so separate out the probe function to simplify the code (just like the codec probe). Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-core.c | 51 +++++++++++++++++++++++++++++++++++-------------- 1 files changed, 36 insertions(+), 15 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 32bc503..d49f0ed 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -986,6 +986,39 @@ err_probe: return ret; } +static int soc_probe_platform(struct snd_soc_card *card, + struct snd_soc_platform *platform) +{ + int ret = 0; + const struct snd_soc_platform_driver *driver = platform->driver; + + platform->card = card; + + if (!try_module_get(platform->dev->driver->owner)) + return -ENODEV; + + if (driver->probe) { + ret = driver->probe(platform); + if (ret < 0) { + dev_err(platform->dev, + "asoc: failed to probe platform %s: %d\n", + platform->name, ret); + goto err_probe; + } + } + + /* mark platform as probed and add to card platform list */ + platform->probed = 1; + list_add(&platform->card_list, &card->platform_dev_list); + + return 0; + +err_probe: + module_put(platform->dev->driver->owner); + + return ret; +} + static void rtd_release(struct device *dev) {} static int soc_post_component_init(struct snd_soc_card *card, @@ -1109,21 +1142,9 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) /* probe the platform */ if (!platform->probed && platform->driver->probe_order == order) { - if (!try_module_get(platform->dev->driver->owner)) - return -ENODEV; - - if (platform->driver->probe) { - ret = platform->driver->probe(platform); - if (ret < 0) { - printk(KERN_ERR "asoc: failed to probe platform %s\n", - platform->name); - module_put(platform->dev->driver->owner); - return ret; - } - } - /* mark platform as probed and add to card platform list */ - platform->probed = 1; - list_add(&platform->card_list, &card->platform_dev_list); + ret = soc_probe_platform(card, platform); + if (ret < 0) + return ret; } /* probe the CODEC DAI */ -- 1.7.4.1

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[alsa-devel] Roland UM-ONE support
by xl600＠teamspode.org 02 Jul '11

02 Jul '11

I am in need of adding UM-ONE support. I believe this interface is simply a repackaged UM-1G, but ALSA 1.0.17 simply won't recognize it. Being new to this level of development, any clues as to how to add this interface support would be welcome. I'd first like to start off by assuming it will work as a UM-1G. If it doesn't, then I'll have to dig deeper. Thanks, Robert

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Re: [alsa-devel] Bug#631963: linux-image-3.0.0-rc4-686-pae: no sound out the rear plugs
by Ben Hutchings 01 Jul '11

01 Jul '11

On Fri, Jul 01, 2011 at 11:06:56PM +0200, Toni Mueller wrote: > > Hi Takashi, > > On Thu, 30.06.2011 at 07:59:19 +0200, Takashi Iwai <tiwai(a)suse.de> wrote: > > I guess this must be the new feature, "Auto-Mute Mode". > > it turns out that you're right - I just had the opportunity to verify > that. Can we consider this closed, then? Or should that feature be disabled by default? Ben. -- Ben Hutchings We get into the habit of living before acquiring the habit of thinking. - Albert Camus

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[alsa-devel] PCM DMA only can fire ISR when it receives 4 samples more than the period, from McBSP port?
by Philip Chu 01 Jul '11

01 Jul '11

HI, I got a problem with my audio capture using McBSP 3 port on OMAP 3530. Linux kernel 2.6.32. For the very first audio data capture, when I expect to receive only one exact period of data through McBSP, PCM DMA ISR never comes. When I check the DMA pointer by using snd_pcm_update_hw_ptr_pos(), I found it was just trash value. If I ask the data sender to send 4 samples more than the period defined, DMA ISR gets fired, the it points to the exact period position, and I assume the extra 4 bytes just get lost? My McBSP is configured as a slave which needs the sender to provide Frame Sync, clocks etc, and those signals are designed to just kick in at the same time. Do the clock signal need to be present earlier than the Frame Sync? Anybody knows what the issue could be? Thanks in advance. Philip Chu

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Re: [alsa-devel] [PATCH 2/5] ASoC: atmel-ssc: add phybase in device structure
by Mark Brown 01 Jul '11

01 Jul '11

On Wed, Jun 29, 2011 at 08:00:16PM +0200, Nicolas Ferre wrote: > struct ssc_device { > struct list_head list; > + resource_size_t phybase; The type here looks rather wrong?

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[alsa-devel] alsa-lib rate plugins - float samples
by Pavel Hofman 01 Jul '11

01 Jul '11

Hi, This is probably a question for Jaroslav Kysela. The better-quality rate plugins offer float resolution, but currently we are deliberately loosing information by down-converting to s16 in plugin_rate API (convert_s16). In rate_samplerate.c, the samples are up-converted to float before calling the libsamplerate API. Speex offers float API too, while we are using the int16 one. There are two ways to handle this: A. Implementing the generic "convert" API function for these plugins, handling the sample - float conversion in each plugin individually. B. Extending the pcm_rate API (pcm_rate.h) with a new optional function convert_float. Since the conversion from general sample format to float is not trivial (using the plugin_ops.h operations), I would be inclined to suggest the extension (well I have it hacked already :-) ). On the other hand, the conversion routine can be located in some common code shared by all the plugins involved and we could keep the existing API, using the generic convert function. Thanks a lot for suggestions. Best regards, Pavel.

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Re: [alsa-devel] [PATCH 1/5] ASoC: wm8731: rework power management
by Mark Brown 01 Jul '11

01 Jul '11

On Wed, Jun 29, 2011 at 08:00:15PM +0200, Nicolas Ferre wrote: Don't mix multiple changes into a single patch! There's no perceptible code overlap between these so I don't understand why you've merged them, it just makes review harder and the changelog less descriptive. > - preserve crystal oscillator across suspend/resume sequence: > enabled by default, it should be kept enabled on resume. This isn't what your code does... > - if codec is in active state: set the active bit at resume time. Your changelog doesn't explain what the actual problem is here... I can see it by studying the diff but the changelog makes me think the change is bad as the subsystem shuts everything down before it enters CODEC suspend... > snd_soc_write(codec, WM8731_ACTIVE, 0x0); > - snd_soc_write(codec, WM8731_PWR, 0xffff); > + /* standby: keep crystal oscillator enabled */ > + snd_soc_write(codec, WM8731_PWR, 0x00df); This doesn't keep the crystal oscillator enabled, this forces it on in suspend (and without looking at the datasheet it also changes way more than the one register bit I'd expect to be changed). If the system isn't using the oscillator then that's not good. I'd expect to see a change to using snd_soc_update_bits() based on your description, or more likely something more involved. > @@ -533,6 +534,8 @@ static int wm8731_suspend(struct snd_soc_codec *codec, pm_message_t state) > static int wm8731_resume(struct snd_soc_codec *codec) > { > wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY); > + if (codec->active) > + snd_soc_write(codec, WM8731_ACTIVE, 0x0001); This should be handled as part of the bias level managemnent and cache sync, if the disable happens in bias level management so should the restore.

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