January 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] wm8974 and sam9g45 atmel
by Michele Da Rold 11 Jul '14

11 Jul '14

Hello everybody, I'm starting a new development with a Atmel SAM9G45 and a WM8974. I'm using kernel 2.6.33.2 I have connected the SSC1 pin TD,TF,TK and the RD pin, so I want to use WM8974 like a slave. I wrote a interface driver based on sam9263_wm8731.c The driver seem to load ok, I recognize the codec but when I start a mp3 playing (I use mpg123) the playback not start. Putting some dedug messagge the driver not call the function "set_bias_level", and call the mute function with mute=0. Testing the MCLK signal I not see any clock on pin because I never call clk_enable!! There is a way to identify the problem? Someone could help me? Thank you in advance Michele Da Rold I put the code of driver (don't care about printk I used it for debug purpose!!) /* * sam9g45_wm8974 -- SoC audio for AT91SAM9G20-based * ATMEL AT91SAM9G20ek board. * * Copyright (C) 2005 SAN People * Copyright (C) 2008 Atmel * * Authors: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> * * Based on sam9263_wm8731.c. * Frank Mandarino <fmandarino(a)endrelia.com> * Copyright 2006 Endrelia Technologies Inc. * Based on corgi.c by: * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * * 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/module.h> #include <linux/moduleparam.h> #include <linux/kernel.h> #include <linux/clk.h> #include <linux/timer.h> #include <linux/interrupt.h> #include <linux/platform_device.h> #include <linux/i2c.h> #include <linux/atmel-ssc.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/soc-dapm.h> #include <asm/mach-types.h> #include <mach/hardware.h> #include <mach/gpio.h> #include "../codecs/wm8974.h" #include "atmel-pcm.h" #include "atmel_ssc_dai.h" #define MCLK_RATE 12000000 /* * As shipped the board does not have inputs. However, it is relatively * straightforward to modify the board to hook them up so support is left * in the driver. */ #undef ENABLE_MIC_INPUT #define ENABLE_MIC_INPUT 1 static struct clk *mclk; static int at91sam9g45ek_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int ret; printk(KERN_DEBUG "at91sam9g45_wm8974 " ": at91sam9g45ek_hw_params() called\n"); /* set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS/*SND_SOC_DAIFMT_CBM_CFM*/); if (ret < 0) return ret; /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS/*SND_SOC_DAIFMT_CBM_CFM*/); if (ret < 0) return ret; return 0; } static struct snd_soc_ops at91sam9g45ek_ops = { .hw_params = at91sam9g45ek_hw_params, }; static int at91sam9g45ek_set_bias_level(struct snd_soc_card *card, enum snd_soc_bias_level level) { static int mclk_on; int ret = 0; printk(KERN_DEBUG "at91sam9g45_wm8974 " ": at91sam9g45ek_set_bias_level() called %d\n",level); switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: if (!mclk_on) ret = clk_enable(mclk); if (ret == 0) mclk_on = 1; break; case SND_SOC_BIAS_OFF: case SND_SOC_BIAS_STANDBY: if (mclk_on) clk_disable(mclk); mclk_on = 0; break; } return ret; } #if 0 static const struct snd_soc_dapm_widget at91sam9g45ek_dapm_widgets[] = { SND_SOC_DAPM_MIC("Int Mic", NULL), SND_SOC_DAPM_SPK("Ext Spk", NULL), }; static const struct snd_soc_dapm_route intercon[] = { /* Speaker output mixer */ {"Speaker Mixer", "PCM Playback Switch", "DAC"}, {"Speaker Mixer", "Aux Playback Switch", "Aux Input"}, {"Speaker Mixer", "Line Bypass Switch", "Boost Mixer"}, /* speaker connected to SPKOUTP */ {"SpkN Out", NULL, "Speaker Mixer"}, {"SpkP Out", NULL, "Speaker Mixer"}, {"SPKOUTN", NULL, "SpkN Out"}, {"SPKOUTP", NULL, "SpkP Out"}, /* Input PGA */ {"Input PGA", "Aux Switch", "Aux Input"}, {"Input PGA", "MicN Switch", "MICN"}, {"Input PGA", "MicP Switch", "MICP"}, }; /* * Logic for a wm974 as connected on a at91sam9g45ek board. */ static int at91sam9g45ek_wm8974_init(struct snd_soc_codec *codec) { struct snd_soc_dai *codec_dai = &codec->dai[0]; int ret; printk(KERN_DEBUG "at91sam9g45_wm8974 " ": at91sam9g45_wm8974_init() called\n"); /* ret = snd_soc_dai_set_sysclk(codec_dai, 0, MCLK_RATE, SND_SOC_CLOCK_IN); if (ret < 0) { printk(KERN_ERR "Failed to set WM8974 SYSCLK: %d\n", ret); return ret; } */ /* Add specific widgets */ snd_soc_dapm_new_controls(codec, at91sam9g45ek_dapm_widgets, ARRAY_SIZE(at91sam9g45ek_dapm_widgets)); /* Set up specific audio path interconnects */ snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); /* not connected */ snd_soc_dapm_nc_pin(codec, "MONOOUT"); snd_soc_dapm_nc_pin(codec, "AUX"); #ifdef ENABLE_MIC_INPUT snd_soc_dapm_enable_pin(codec, "Int Mic"); #else snd_soc_dapm_nc_pin(codec, "Int Mic"); #endif /* always connected */ snd_soc_dapm_enable_pin(codec, "Ext Spk"); snd_soc_dapm_sync(codec); return 0; } #endif static struct snd_soc_dai_link at91sam9g45ek_dai = { .name = "WM8974", .stream_name = "WM8974 PCM", .cpu_dai = &atmel_ssc_dai[1], .codec_dai = &wm8974_dai, #if 0 .init = at91sam9g45ek_wm8974_init, #endif .ops = &at91sam9g45ek_ops, }; static struct snd_soc_card snd_soc_at91sam9g45ek = { .name = "AT91SAMG45", .platform = &atmel_soc_platform, .dai_link = &at91sam9g45ek_dai, .num_links = 1, .set_bias_level = at91sam9g45ek_set_bias_level, }; static struct snd_soc_device at91sam9g45ek_snd_devdata = { .card = &snd_soc_at91sam9g45ek, .codec_dev = &soc_codec_dev_wm8974, }; static struct platform_device *at91sam9g45ek_snd_device; static int __init at91sam9g45ek_init(void) { struct atmel_ssc_info *ssc_p = at91sam9g45ek_dai.cpu_dai->private_data; struct ssc_device *ssc = NULL; struct clk *main; int ret; at91_set_A_periph(AT91_PIN_PD27, 0); /* * Codec MCLK is supplied by PCK1 - set it up. */ mclk = clk_get(NULL, "pck1"); if (IS_ERR(mclk)) { printk(KERN_ERR "ASoC: Failed to get MCLK\n"); ret = PTR_ERR(mclk); goto err; } main = clk_get(NULL, "main"); if (IS_ERR(main)) { printk(KERN_ERR "ASoC: Failed to get PLLB\n"); ret = PTR_ERR(mclk); goto err_mclk; } ret = clk_set_parent(mclk, main); clk_put(main); if (ret != 0) { printk(KERN_ERR "ASoC: Failed to set MCLK parent\n"); goto err_mclk; } clk_set_rate(mclk, MCLK_RATE); /* * Request SSC device */ ssc = ssc_request(1); if (IS_ERR(ssc)) { printk(KERN_ERR "ASoC: Failed to request SSC 1\n"); ret = PTR_ERR(ssc); ssc = NULL; goto err_ssc; } ssc_p->ssc = ssc; at91sam9g45ek_snd_device = platform_device_alloc("soc-audio", -1); if (!at91sam9g45ek_snd_device) { printk(KERN_ERR "ASoC: Platform device allocation failed\n"); ret = -ENOMEM; } platform_set_drvdata(at91sam9g45ek_snd_device, &at91sam9g45ek_snd_devdata); at91sam9g45ek_snd_devdata.dev = &at91sam9g45ek_snd_device->dev; ret = platform_device_add(at91sam9g45ek_snd_device); if (ret) { printk(KERN_ERR "ASoC: Platform device allocation failed\n"); platform_device_put(at91sam9g45ek_snd_device); } return ret; err_ssc: ssc_free(ssc); ssc_p->ssc = NULL; err_mclk: clk_put(mclk); mclk = NULL; err: return ret; } static void __exit at91sam9g45ek_exit(void) { struct atmel_ssc_info *ssc_p = at91sam9g45ek_dai.cpu_dai->private_data; struct ssc_device *ssc; if (ssc_p != NULL) { ssc = ssc_p->ssc; if (ssc != NULL) ssc_free(ssc); ssc_p->ssc = NULL; } platform_device_unregister(at91sam9g45ek_snd_device); at91sam9g45ek_snd_device = NULL; clk_put(mclk); mclk = NULL; } module_init(at91sam9g45ek_init); module_exit(at91sam9g45ek_exit); /* Module information */ MODULE_AUTHOR("Sedji Gaouaou <sedji.gaouaou(a)atmel.com>"); MODULE_DESCRIPTION("ALSA SoC AT91SAM9G45_WM8974"); MODULE_LICENSE("GPL");

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[alsa-devel] [RFC][PATCH v1] ASoC: fsl_ssi: Add DAI master mode support for SSI on i.MX series
by Nicolin Chen 13 Jun '14

13 Jun '14

From: Nicolin Chen <b42378(a)freescale.com> This patch adds three main functions for DAI master mode: set_dai_fmt(), set_dai_sysclk() and set_dai_tdm_slot(), and one essential baud clock accordingly. After appending this patch, the fsl_ssi driver on i.MX series has the ability to derive LRCLK and BCLK from baud clock source so as to support some audio Codecs which can only be used in slave mode. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 280 +++++++++++++++++++++++++++++++++++++++++++++++- sound/soc/fsl/fsl_ssi.h | 2 + 2 files changed, 278 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index f9f4569..b2ebaf8 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -38,6 +38,7 @@ #include <linux/device.h> #include <linux/delay.h> #include <linux/slab.h> +#include <linux/spinlock.h> #include <linux/of_address.h> #include <linux/of_irq.h> #include <linux/of_platform.h> @@ -139,7 +140,10 @@ struct fsl_ssi_private { bool ssi_on_imx; bool imx_ac97; bool use_dma; + bool baudclk_locked; u8 i2s_mode; + spinlock_t baudclk_lock; + struct clk *baudclk; struct clk *clk; struct snd_dmaengine_dai_dma_data dma_params_tx; struct snd_dmaengine_dai_dma_data dma_params_rx; @@ -434,13 +438,18 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai); + unsigned long flags; /* First, we only do fsl_ssi_setup() when SSI is going to be active. * Second, fsl_ssi_setup was already called by ac97_init earlier if * the driver is in ac97 mode. */ - if (!dai->active && !ssi_private->imx_ac97) + if (!dai->active && !ssi_private->imx_ac97) { fsl_ssi_setup(ssi_private); + spin_lock_irqsave(&ssi_private->baudclk_lock, flags); + ssi_private->baudclk_locked = false; + spin_unlock_irqrestore(&ssi_private->baudclk_lock, flags); + } return 0; } @@ -502,6 +511,243 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, } /** + * fsl_ssi_set_dai_fmt - configure Digital Audio Interface Format. + */ +static int fsl_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) +{ + struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + u32 strcr = 0, stcr, srcr, scr, mask; + + scr = read_ssi(&ssi->scr) & ~(CCSR_SSI_SCR_SYN | CCSR_SSI_SCR_I2S_MODE_MASK); + scr |= CCSR_SSI_SCR_NET; + + mask = CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFDIR | CCSR_SSI_STCR_TXDIR | + CCSR_SSI_STCR_TSCKP | CCSR_SSI_STCR_TFSI | CCSR_SSI_STCR_TFSL | + CCSR_SSI_STCR_TEFS; + stcr = read_ssi(&ssi->stcr) & ~mask; + srcr = read_ssi(&ssi->srcr) & ~mask; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + ssi_private->i2s_mode = CCSR_SSI_SCR_I2S_MODE_MASTER; + break; + case SND_SOC_DAIFMT_CBM_CFM: + ssi_private->i2s_mode = CCSR_SSI_SCR_I2S_MODE_SLAVE; + break; + default: + return -EINVAL; + } + scr |= ssi_private->i2s_mode; + + /* Data on rising edge of bclk, frame low, 1clk before data */ + strcr |= CCSR_SSI_STCR_TFSI | CCSR_SSI_STCR_TSCKP | + CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TEFS; + break; + case SND_SOC_DAIFMT_LEFT_J: + /* Data on rising edge of bclk, frame high */ + strcr |= CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TSCKP; + break; + case SND_SOC_DAIFMT_DSP_A: + /* Data on rising edge of bclk, frame high, 1clk before data */ + strcr |= CCSR_SSI_STCR_TFSL | CCSR_SSI_STCR_TSCKP | + CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TEFS; + break; + case SND_SOC_DAIFMT_DSP_B: + /* Data on rising edge of bclk, frame high */ + strcr |= CCSR_SSI_STCR_TFSL | CCSR_SSI_STCR_TSCKP | + CCSR_SSI_STCR_TXBIT0; + break; + default: + return -EINVAL; + } + + /* DAI clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + /* Nothing to do for both normal cases */ + break; + case SND_SOC_DAIFMT_IB_NF: + /* Invert bit clock */ + strcr ^= CCSR_SSI_STCR_TSCKP; + break; + case SND_SOC_DAIFMT_NB_IF: + /* Invert frame clock */ + strcr ^= CCSR_SSI_STCR_TFSI; + break; + case SND_SOC_DAIFMT_IB_IF: + /* Invert both clocks */ + strcr ^= CCSR_SSI_STCR_TSCKP; + strcr ^= CCSR_SSI_STCR_TFSI; + break; + default: + return -EINVAL; + } + + /* DAI clock master masks */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + strcr |= CCSR_SSI_STCR_TFDIR | CCSR_SSI_STCR_TXDIR; + scr |= CCSR_SSI_SCR_SYS_CLK_EN; + break; + case SND_SOC_DAIFMT_CBM_CFM: + scr &= ~CCSR_SSI_SCR_SYS_CLK_EN; + break; + default: + return -EINVAL; + } + + stcr |= strcr; + srcr |= strcr; + + if (ssi_private->cpu_dai_drv.symmetric_rates) { + /* Need to clear RXDIR when using SYNC mode */ + srcr &= ~CCSR_SSI_SRCR_RXDIR; + scr |= CCSR_SSI_SCR_SYN; + } + + write_ssi(stcr, &ssi->stcr); + write_ssi(srcr, &ssi->srcr); + write_ssi(scr, &ssi->scr); + + return 0; +} + +/** + * fsl_ssi_set_dai_sysclk - configure Digital Audio Interface bit clock + * + * Note: This function can be only called when using SSI as DAI master + * + * Quick instruction for parameters: + * freq: Output BCLK frequency = samplerate * 32 (fixed) * channels + * dir: SND_SOC_CLOCK_OUT -> TxBCLK, SND_SOC_CLOCK_IN -> RxBCLK. + */ +static int fsl_ssi_set_dai_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + int synchronous = ssi_private->cpu_dai_drv.symmetric_rates, ret; + u32 pm = 999, div2, psr, stccr, mask, afreq, factor, i; + unsigned long flags, clkrate, baudrate, tmprate; + u64 sub, savesub = 100000; + + /* Don't apply it to any non-baudclk circumstance */ + if (IS_ERR(ssi_private->baudclk)) + return -EINVAL; + + /* It should be already enough to divide clock by setting pm alone */ + psr = 0; + div2 = 0; + + factor = (div2 + 1) * (7 * psr + 1) * 2; + + for (i = 0; i < 255; i++) { + /* The bclk rate must be smaller than 1/5 sysclk rate */ + if (factor * (i + 1) < 5) + continue; + + tmprate = freq * factor * (i + 2); + clkrate = clk_round_rate(ssi_private->baudclk, tmprate); + + do_div(clkrate, factor); + afreq = (u32)clkrate / (i + 1); + + if (freq == afreq) + sub = 0; + else if (freq / afreq == 1) + sub = freq - afreq; + else if (afreq / freq == 1) + sub = afreq - freq; + else + continue; + + /* Calculate the fraction */ + sub *= 100000; + do_div(sub, freq); + + if (sub < savesub) { + baudrate = tmprate; + savesub = sub; + pm = i; + } + + /* We are lucky */ + if (savesub == 0) + break; + } + + /* No proper pm found if it is still remaining the initial value */ + if (pm == 999) { + dev_err(cpu_dai->dev, "failed to handle the required sysclk\n"); + return -EINVAL; + } + + stccr = CCSR_SSI_SxCCR_PM(pm + 1) | (div2 ? CCSR_SSI_SxCCR_DIV2 : 0) | + (psr ? CCSR_SSI_SxCCR_PSR : 0); + mask = CCSR_SSI_SxCCR_PM_MASK | CCSR_SSI_SxCCR_DIV2 | CCSR_SSI_SxCCR_PSR; + + if (dir == SND_SOC_CLOCK_OUT || synchronous) + write_ssi_mask(&ssi->stccr, mask, stccr); + else + write_ssi_mask(&ssi->srccr, mask, stccr); + + spin_lock_irqsave(&ssi_private->baudclk_lock, flags); + if (!ssi_private->baudclk_locked) { + ret = clk_set_rate(ssi_private->baudclk, baudrate); + if (ret) { + spin_unlock_irqrestore(&ssi_private->baudclk_lock, flags); + dev_err(cpu_dai->dev, "failed to set baudclk rate\n"); + return -EINVAL; + } + ssi_private->baudclk_locked = true; + } + spin_unlock_irqrestore(&ssi_private->baudclk_lock, flags); + + return 0; +} + +/** + * fsl_ssi_set_dai_tdm_slot - set TDM slot number + * + * Note: This function can be only called when using SSI as DAI master + */ +static int fsl_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask, + u32 rx_mask, int slots, int slot_width) +{ + struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + u32 val; + + /* The slot number should be >= 2 if using Network mode or I2S mode */ + val = read_ssi(&ssi->scr) & (CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_NET); + if (val && slots < 2) { + dev_err(cpu_dai->dev, "slot number should be >= 2 in I2S or NET\n"); + return -EINVAL; + } + + write_ssi_mask(&ssi->stccr, CCSR_SSI_SxCCR_DC_MASK, + CCSR_SSI_SxCCR_DC(slots)); + write_ssi_mask(&ssi->srccr, CCSR_SSI_SxCCR_DC_MASK, + CCSR_SSI_SxCCR_DC(slots)); + + /* The register SxMSKs needs SSI to provide essential clock due to + * hardware design. So we here temporarily enable SSI to set them. + */ + val = read_ssi(&ssi->scr) & CCSR_SSI_SCR_SSIEN; + write_ssi_mask(&ssi->scr, 0, CCSR_SSI_SCR_SSIEN); + + write_ssi(tx_mask, &ssi->stmsk); + write_ssi(rx_mask, &ssi->srmsk); + + write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, val); + + return 0; +} + +/** * fsl_ssi_trigger: start and stop the DMA transfer. * * This function is called by ALSA to start, stop, pause, and resume the DMA @@ -517,6 +763,7 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai); struct ccsr_ssi __iomem *ssi = ssi_private->ssi; unsigned int sier_bits; + unsigned long flags; /* * Enable only the interrupts and DMA requests @@ -557,8 +804,12 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_RE, 0); if (!ssi_private->imx_ac97 && (read_ssi(&ssi->scr) & - (CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE)) == 0) + (CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE)) == 0) { write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0); + spin_lock_irqsave(&ssi_private->baudclk_lock, flags); + ssi_private->baudclk_locked = false; + spin_unlock_irqrestore(&ssi_private->baudclk_lock, flags); + } break; default: @@ -585,6 +836,9 @@ static int fsl_ssi_dai_probe(struct snd_soc_dai *dai) static const struct snd_soc_dai_ops fsl_ssi_dai_ops = { .startup = fsl_ssi_startup, .hw_params = fsl_ssi_hw_params, + .set_fmt = fsl_ssi_set_dai_fmt, + .set_sysclk = fsl_ssi_set_dai_sysclk, + .set_tdm_slot = fsl_ssi_set_dai_tdm_slot, .trigger = fsl_ssi_trigger, }; @@ -897,6 +1151,9 @@ static int fsl_ssi_probe(struct platform_device *pdev) /* Older 8610 DTs didn't have the fifo-depth property */ ssi_private->fifo_depth = 8; + ssi_private->baudclk_locked = false; + spin_lock_init(&ssi_private->baudclk_lock); + if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx21-ssi")) { u32 dma_events[2]; ssi_private->ssi_on_imx = true; @@ -914,6 +1171,15 @@ static int fsl_ssi_probe(struct platform_device *pdev) goto error_irqmap; } + /* For those SLAVE implementations, we ingore non-baudclk cases + * and, instead, abandon MASTER mode that needs baud clock. + */ + ssi_private->baudclk = devm_clk_get(&pdev->dev, "baud"); + if (IS_ERR(ssi_private->baudclk)) + dev_warn(&pdev->dev, "could not get baud clock: %d\n", ret); + else + clk_prepare_enable(ssi_private->baudclk); + /* * We have burstsize be "fifo_depth - 2" to match the SSI * watermark setting in fsl_ssi_startup(). @@ -1059,8 +1325,11 @@ error_dev: device_remove_file(&pdev->dev, dev_attr); error_clk: - if (ssi_private->ssi_on_imx) + if (ssi_private->ssi_on_imx) { + if (!IS_ERR(ssi_private->baudclk)) + clk_disable_unprepare(ssi_private->baudclk); clk_disable_unprepare(ssi_private->clk); + } error_irqmap: irq_dispose_mapping(ssi_private->irq); @@ -1076,8 +1345,11 @@ static int fsl_ssi_remove(struct platform_device *pdev) platform_device_unregister(ssi_private->pdev); snd_soc_unregister_component(&pdev->dev); device_remove_file(&pdev->dev, &ssi_private->dev_attr); - if (ssi_private->ssi_on_imx) + if (ssi_private->ssi_on_imx) { + if (!IS_ERR(ssi_private->baudclk)) + clk_disable_unprepare(ssi_private->baudclk); clk_disable_unprepare(ssi_private->clk); + } irq_dispose_mapping(ssi_private->irq); return 0; diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h index e6b9a69..e6b6324 100644 --- a/sound/soc/fsl/fsl_ssi.h +++ b/sound/soc/fsl/fsl_ssi.h @@ -125,7 +125,9 @@ struct ccsr_ssi { #define CCSR_SSI_SRCR_REFS 0x00000001 /* STCCR and SRCCR */ +#define CCSR_SSI_SxCCR_DIV2_SHIFT 18 #define CCSR_SSI_SxCCR_DIV2 0x00040000 +#define CCSR_SSI_SxCCR_PSR_SHIFT 17 #define CCSR_SSI_SxCCR_PSR 0x00020000 #define CCSR_SSI_SxCCR_WL_SHIFT 13 #define CCSR_SSI_SxCCR_WL_MASK 0x0001E000 -- 1.8.4

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[alsa-devel] Segfaults in snd_pcm_area_silence() when using softvol
by Tanu Kaskinen 18 May '14

18 May '14

Hi, A bug was reported to pulseaudio about a segfault in alsa-lib code when using softvol: https://bugs.freedesktop.org/show_bug.cgi?id=51060 . It has been observed by several people, and this mail was prompted by the last bug comment, which rightly recommends getting in touch with the alsa developers: > I've had this problem too and the problem only occurs with > -ftree-vectorize. > > It looks to me as if this is a gcc bug, as it produces a "movdqa" or > "vmovdqa" instruction which is used with an unaligned pointer. > > Now, I have no clue why gcc assumes the pointer would be an aligned > one, maybe it is even right and something else is wrong, but I haven't > spent too much time digging into this. > > (note that it still happens with gcc 4.7 and the current 4.8 > development one) > > Also, this bug should probably be brought up with the alsa guys as > well, as I don't think pulseaudio is at fault here. Here's the stack trace for your convenience (see the linked bug for more details) #0 snd_pcm_area_silence (dst_area=0x7fffe875bb00, dst_offset=<optimized out>, samples=2, format=SND_PCM_FORMAT_S16_LE) at pcm.c:2549 #1 0x00007fffe87b4398 in snd_pcm_areas_silence (dst_areas=0x5555557fa100, dst_offset=891, channels=2, frames=439, format=SND_PCM_FORMAT_S16_LE) at pcm.c:2662 #2 0x00007fffe87f9bdb in softvol_convert_stereo_vol (svol=0x555555813cf0, dst_areas=0x5555557fa0e0, dst_offset=891, src_areas=0x5555557fa0e0, src_offset=<optimized out>, channels=2, frames=439) at pcm_softvol.c:286 #3 0x00007fffe87fae58 in snd_pcm_softvol_write_areas (pcm=0x55555581db60, areas=0x5555557fa0e0, offset=891, size=439, slave_areas=0x5555557fa0e0, slave_offset=891, slave_sizep=0x7fffe875bcd0) at pcm_softvol.c:607 #4 0x00007fffe87c6fc7 in snd_pcm_plugin_mmap_commit (pcm=0x55555581db60, offset=<optimized out>, size=439) at pcm_plugin.c:424 #5 0x00007fffe87b9676 in snd_pcm_mmap_commit (pcm=<optimized out>, offset=<optimized out>, frames=<optimized out>) at pcm.c:6584 #6 0x00007fffe8a74876 in mmap_write (on_timeout=false, polled=false, sleep_usec=<synthetic pointer>, u=0x5555557e6d40) at modules/alsa/alsa-sink.c:636 #7 thread_func (userdata=0x5555557e6d40) at modules/alsa/alsa-sink.c:1694 #8 0x00007ffff722c039 in internal_thread_func (userdata=0x55555587ef80) at pulsecore/thread-posix.c:83 #9 0x00007ffff4a24e2c in ?? () from /lib64/libpthread.so.0 #10 0x00007ffff3ccc1dd in clone () from /lib64/libc.so.6 -- Tanu

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[alsa-devel] [PATCH v3 0/5] ALSA: hda - hdmi: ATI/AMD multi-channel and HBR support
by Anssi Hannula 14 May '14

14 May '14

Hi all! Here's another revision of the ATI/AMD multichannel+HBR patchset. Changes: - added customizable callbacks instead of polluting generic code - fixed wrong codec id in rev3+ check (as previously noted) - cosmetics I've only tested this on NVIDIA/Intel, but I'd guess at least Peter should be able to test this on AMD in a few days or so. The earlier revisions were tested to fully work. Combined patch can be found at: http://onse.fi/files/atihdmi5.patch However, this patchset is on top recent sound-next commits so it does not directly apply on older kernels. Olivier, I hope you will be able to test this at some point as well (same things as last time), even though it will be a bit more work :) Anssi Hannula (5): ALSA: hda - hdmi: Allow HDA patches to customize more operations ALSA: hda - hdmi: Add ATI/AMD multi-channel audio support ALSA: hda - hdmi: Add ELD emulation for ATI/AMD codecs ALSA: hda - hdmi: Add HBR bitstreaming support for ATI/AMD HDMI codecs ALSA: hda - hdmi: Disable ramp-up/down for non-PCM on AMD codecs sound/pci/hda/hda_eld.c | 151 +++++++++++ sound/pci/hda/hda_local.h | 4 + sound/pci/hda/patch_hdmi.c | 643 +++++++++++++++++++++++++++++++++++++-------- 3 files changed, 689 insertions(+), 109 deletions(-) -- Anssi Hannula

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[alsa-devel] Tenor TE8802, USB : clics and crackles during music play
by Eric F 07 Apr '14

07 Apr '14

Hello, Been searching for days on Internet for this issue, without luck :( I own a Teac UD-H01 : when connected via USB => crackles and clicks

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[alsa-devel] [PATCH] ALSA: usb-audio - Capture and duplex support for Digidesign Mbox 1 sound card.
by Damien Zammit 01 Apr '14

01 Apr '14

This patch completes the support for Digidesign Mbox 1 sound card by adding capture. The issue with this card was that it has two endpoints on the same interface. This patch creates a dual endpoint quirk. The quirk interface needs a second audioformat struct for this to work which I called ".data2". Signed-off-by: Damien Zammit <damien(a)zamaudio.com>

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[alsa-devel] Fwd: Re: no reset_resume for driver snd-usb-audio for logitech headset H600
by baumber＠hotmail.com 26 Mar '14

26 Mar '14

FWD: Any news to the snd-usb-audio issue? Thanks, Bernhard Old Message: ------------- I have tested the patch with kernel 3.12.0 with suspend and hibernate, and the problem is gone. No messages "no reset_resume for driver snd-usb-audio?" or other negative effects/messages. After resume, the usb_audio_device (Headset) is the master channel in KDE Mixer, as it should be. => So the problem is solved for me. Can you please add the patch to the latest kernels and LTS-kernels or whitelist the workaround for my device (ID 046d:0a29 Logitech, Inc. H600 [Wireless Headset]) Thanks, Bernhard On 2013-10-20 23:35, Takashi Iwai wrote: > At Sun, 20 Oct 2013 23:18:13 +0200, > <baumber(a)hotmail.com> wrote: >> >> On 2013-10-09 11:16 (CEST,UTC+2), Takashi Iwai wrote: >>>>>> diff --git a/sound/usb/card.c b/sound/usb/card.c >>>>>> index 9d9de8d..c591a83 100644 >>>>>> --- a/sound/usb/card.c >>>>>> +++ b/sound/usb/card.c >>>>>> @@ -747,6 +747,7 @@ static struct usb_driver usb_audio_driver = { >>>>>> .disconnect = usb_audio_disconnect, >>>>>> .suspend = usb_audio_suspend, >>>>>> .resume = usb_audio_resume, >>>>>> + .reset_resume = usb_audio_resume, >>>>>> .id_table = usb_audio_ids, >>>>>> .supports_autosuspend = 1, >>>>>> }; >> >> Shall I add the line in b/sound/usb/card.c and try it? Which kernel version shall I use? > > It doesn't matter which kernel version is. > The error message should disappear, but I don't know whether any other > positive effect you get. As Clemens pointed, there can be some > negative effects by this, so it cannot be taken as is. But at least, > it'd be good to know the result... > > > thanks, > > Takashi

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[alsa-devel] Ambiguous DAPM widget names and DPCM
by Daniel Mack 21 Mar '14

21 Mar '14

Hi Mark, Hi Liam, I spent some hours reverse-engineering the DPCM code, and I plan to prepare a patch to add some documentation once I've assembled a fully working setup. However, I have a general question regarding DAPM widget names. DPCM uses DAPM widgets and routes in order to determine the FE <-> BE connections at runtime. The problem with this is that widget names are very ambiguous, as the automatically created widgets for DAIs are named after their stream_name properties, which contain "Playback" or "Capture" for all components in my case. Hence, the code which walks the connections (snd_soc_dapm_dai_get_connected_widgets()) will most likely match the wrong one (which has empty sound/sink lists), which results in a failure like this: [ 16.606268] fe: ASoC: fe no valid playback route I hacked the names of all components so they are unique, but even then, the problem is that snd_soc_dapm_new_dai_widgets() is called twice for the CPU DAI, once from soc_probe_platform(), and later from soc_probe_link_dais(), and both calls would create DAPM widgets with identical names, resulting in the same problem as described above. I don't know how to handle this, but I believe that nobody was really hit by this issue yet, as DPCM doesn't seem to be used widely, at least not by any machine code in mainline. One idea would be to change the automatically generated names of DAI widgets in order to make them unique, or re-factor the lookup routines. Any opinion on that? Do I miss a general consideration here? With more hacks in place, the setup somehow succeeds, but when the stream is opened, we'll hit a BUG() in sound/core/pcm_memory.c, because the FE device's DMA type is undefined. But that's another topic I'll address later. Thanks, Daniel

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[alsa-devel] Regression caused by "ASoC: core: Rework SOC_DOUBLE_R_SX_TLV add SOC_SINGLE_SX_TLV"
by Thomas Petazzoni 19 Mar '14

19 Mar '14

Brian, Liam, Mark, I am currently bringing up audio support on the Marvell Armada 370 ARM SoC, for which the development board uses the CS42L51 audio codec. It turns out that the commit 1d99f2436d0d1c7741d6dfd9d27b5376cdbbca40 ("ASoC: core: Rework SOC_DOUBLE_R_SX_TLV add SOC_SINGLE_SX_TLV") causes a problem here, and reverting this commit gets things back to normal. Basically, when this commit is applied, the "PCM", "Analog" and "ADC Mixer" volume controls in alsamixer cannot be changed to any other value than 60 or 61. The volume control doesn't allow to go above 61 and below 60. If I remember correctly, at the codec level, the values configured for PCM in registers 0x10 and 0x11 are either 0x0 or 0x1 depending on whether you select 60 or 61 in alsamixer. Reverting the commit mentioned above makes alsamixer behave normally: the volume can be adjusted between 0 and 100 as expected, the values in register 0x10 and 0x11 are meaningful, and the behavior is as expected: with a volume to 0, I hear nothing in my headphones, with a volume set to 100, the volume is maximum, with a nice progressive volume increase in-between. Therefore, I believe that the commit has a problem. I haven't yet investigated where the problem is, maybe just the values passed to SOC_DOUBLE_R_SX_TLV(), or maybe in the implementation of the volume control functions themselves. I can certainly start investigating, but maybe the author of the commit will immediately realize where the problem could be. Thanks a lot! Thomas -- Thomas Petazzoni, CTO, Free Electrons Embedded Linux, Kernel and Android engineering http://free-electrons.com

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Re: [alsa-devel] Zoom R16
by Jason Mancine 11 Mar '14

11 Mar '14

Can't seem to compile 3.11 or 3.12 for the distro I am using (AVLInux)... tested it in Ubuntu Studio 13.10 and the quirk didn't seem to have any effect. The output of lsusb is the same, but the device doesn't show up as a sound card. On Sun, Nov 24, 2013 at 9:18 AM, Daniel Mack <daniel(a)zonque.org> wrote: > On Sun, Nov 24, 2013 at 09:03:47AM -0500, Jason Mancine wrote: > > Thanks, I will give it a try. By recent kernel do you mean latest > mainline > > or should my distro's 3.6.11 be new enough? > > Anything >= 3.11 should do. > > In further replies, please do not drop the alsa-devel mailing list. > > > Thanks, > Daniel >

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