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

[alsa-devel] [PATCH v3 0/3] ASoC: omap-mcpdm: New McPDM driver
by Peter Ujfalusi 06 Sep '11

06 Sep '11

Hello, Changes since v2 [1]: - Code for the DAPM_SUPPLY widget attach has been moved to core (Request from Mark, if I understood correctly). - Makes it reusable in other drivers - Additional code still needed in DAI (event handler, selection of the dai), and machine drivers (DAPM routing) - Should work even if we have multiple instances of the DAI (coment from Lars-Peter Clausen) [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2011-August/042923.html Regards, Peter --- Misael Lopez Cruz (1): ASoC: omap-mcpdm: Replace legacy driver Peter Ujfalusi (2): ASoC: DAPM: Add private data pointer for DAPM widget ASoC: soc-dapm: API to attach DAPM_SUPPLY to be used for dai include/sound/soc-dapm.h | 18 + sound/soc/omap/Makefile | 2 +- sound/soc/omap/mcpdm.c | 474 ------------------------- sound/soc/omap/mcpdm.h | 152 -------- sound/soc/omap/omap-mcpdm.c | 813 ++++++++++++++++++++++++++++--------------- sound/soc/omap/omap-mcpdm.h | 97 +++++ sound/soc/omap/sdp4430.c | 14 +- sound/soc/soc-dapm.c | 27 ++ 8 files changed, 690 insertions(+), 907 deletions(-) delete mode 100644 sound/soc/omap/mcpdm.c delete mode 100644 sound/soc/omap/mcpdm.h rewrite sound/soc/omap/omap-mcpdm.c (46%) create mode 100644 sound/soc/omap/omap-mcpdm.h -- 1.7.6.1

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[alsa-devel] [PATCH v2 1/2] ASoC: mxs-saif: add record function
by Dong Aisheng 06 Sep '11

06 Sep '11

1. add different clkmux mode handling for record. SAIF can use two instances to implement full duplex (playback & recording) and record saif may work on EXTMASTER mode that is using other saif's BITCLK&LRCLK. The clkmux mode is determined by saif's platform data and machine specific clkmux setting is done in pdata->init(). 2. support playback and capture simutaneously however the sample rates can not be different due to hw limitation. Signed-off-by: Dong Aisheng <b29396(a)freescale.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg(a)ti.com> Cc: Sascha Hauer <s.hauer(a)pengutronix.de> Cc: Wolfram Sang <w.sang(a)pengutronix.de> --- Changes since v1: * calc the delay based on the rate * change the saif.h #ifndef micro to match the directory --- include/sound/saif.h | 16 +++++ sound/soc/mxs/mxs-saif.c | 166 ++++++++++++++++++++++++++++++++++++++++++---- sound/soc/mxs/mxs-saif.h | 4 + 3 files changed, 172 insertions(+), 14 deletions(-) diff --git a/include/sound/saif.h b/include/sound/saif.h new file mode 100644 index 0000000..9e25b25 --- /dev/null +++ b/include/sound/saif.h @@ -0,0 +1,16 @@ +/* + * Copyright 2011 Freescale Semiconductor, Inc. All Rights Reserved. + * + * 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. + */ + +#ifndef __SOUND_SAIF_H__ +#define __SOUND_SAIF_H__ + +struct mxs_saif_platform_data { + unsigned int clkmux; + int (*init) (void); +}; +#endif diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 530017f..71c6ad2 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -23,19 +23,44 @@ #include <linux/dma-mapping.h> #include <linux/clk.h> #include <linux/delay.h> +#include <linux/time.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/saif.h> #include <mach/dma.h> #include <asm/mach-types.h> #include <mach/hardware.h> #include <mach/mxs.h> +#include <mach/digctl.h> #include "mxs-saif.h" static struct mxs_saif *mxs_saif[2]; +/* + * SAIF is a little different with other normal SOC DAIs on clock using. + * + * For MXS, two SAIF modules are instantiated on-chip. + * Each SAIF has a set of clock pins and can be operating in master + * mode simultaneously if they are connected to different off-chip codecs. + * Also, one of the two SAIFs can master or drive the clock pins while the + * other SAIF, in slave mode, receives clocking from the master SAIF. + * This also means that both SAIFs must operate at the same sample rate. + * Following are the valid configurations for SAIF1 and SAIF2 on the i.MX28 + * + * HW_SAIF_CLKMUX_SEL: + * DIRECT(0x0): SAIF0 clock pins selected for SAIF0 input clocks, and SAIF1 + * clock pins selected for SAIF1 input clocks. + * CROSSINPUT(0x1): SAIF1 clock inputs selected for SAIF0 input clocks, and + * SAIF0 clock inputs selected for SAIF1 input clocks. + * EXTMSTR0(0x2): SAIF0 clock pin selected for both SAIF0 and SAIF1 input + * clocks. + * EXTMSTR1(0x3): SAIF1 clock pin selected for both SAIF0 and SAIF1 input + * clocks. + */ + static int mxs_saif_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { @@ -52,6 +77,36 @@ static int mxs_saif_set_dai_sysclk(struct snd_soc_dai *cpu_dai, } /* + * Since SAIF may work on EXTMASTER mode, IOW, it's working BITCLK&LRCLK + * is provided by other SAIF, we provide a interface here to get its master. + * Note that for DIRECT mode, it's master is itself. + */ +static struct mxs_saif *mxs_saif_get_master(struct mxs_saif * saif) +{ + struct mxs_saif *master_saif; + + /* get master saif according to different clkmux setting */ + switch (saif->clkmux) { + case MXS_DIGCTL_SAIF_CLKMUX_DIRECT: + master_saif = saif; + break; + case MXS_DIGCTL_SAIF_CLKMUX_CROSSINPUT: + master_saif = saif->id ? mxs_saif[1] : mxs_saif[0]; + break; + case MXS_DIGCTL_SAIF_CLKMUX_EXTMSTR0: + master_saif = mxs_saif[0]; + break; + case MXS_DIGCTL_SAIF_CLKMUX_EXTMSTR1: + master_saif = mxs_saif[1]; + break; + default: + return NULL; + } + + return master_saif; +} + +/* * Set SAIF clock and MCLK */ static int mxs_saif_set_clk(struct mxs_saif *saif, @@ -60,8 +115,27 @@ static int mxs_saif_set_clk(struct mxs_saif *saif, { u32 scr; int ret; + struct mxs_saif *master_saif; - scr = __raw_readl(saif->base + SAIF_CTRL); + dev_dbg(saif->dev, "mclk %d rate %d\n", mclk, rate); + + /* Set master saif to generate proper clock */ + master_saif = mxs_saif_get_master(saif); + if (!master_saif) + return -EINVAL; + + dev_dbg(saif->dev, "master saif%d clkmux 0x%x\n", + master_saif->id, saif->clkmux); + + /* Checking if can playback and capture simutaneously */ + if (master_saif->ongoing && rate != master_saif->cur_rate) { + dev_err(saif->dev, + "can not change clock, master saif%d(rate %d) is ongoing\n", + master_saif->id, master_saif->cur_rate); + return -EINVAL; + } + + scr = __raw_readl(master_saif->base + SAIF_CTRL); scr &= ~BM_SAIF_CTRL_BITCLK_MULT_RATE; scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE; @@ -75,27 +149,29 @@ static int mxs_saif_set_clk(struct mxs_saif *saif, * * If MCLK is not used, we just set saif clk to 512*fs. */ - if (saif->mclk_in_use) { + if (master_saif->mclk_in_use) { if (mclk % 32 == 0) { scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE; - ret = clk_set_rate(saif->clk, 512 * rate); + ret = clk_set_rate(master_saif->clk, 512 * rate); } else if (mclk % 48 == 0) { scr |= BM_SAIF_CTRL_BITCLK_BASE_RATE; - ret = clk_set_rate(saif->clk, 384 * rate); + ret = clk_set_rate(master_saif->clk, 384 * rate); } else { /* SAIF MCLK should be either 32x or 48x */ return -EINVAL; } } else { - ret = clk_set_rate(saif->clk, 512 * rate); + ret = clk_set_rate(master_saif->clk, 512 * rate); scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE; } if (ret) return ret; - if (!saif->mclk_in_use) { - __raw_writel(scr, saif->base + SAIF_CTRL); + master_saif->cur_rate = rate; + + if (!master_saif->mclk_in_use) { + __raw_writel(scr, master_saif->base + SAIF_CTRL); return 0; } @@ -137,7 +213,7 @@ static int mxs_saif_set_clk(struct mxs_saif *saif, return -EINVAL; } - __raw_writel(scr, saif->base + SAIF_CTRL); + __raw_writel(scr, master_saif->base + SAIF_CTRL); return 0; } @@ -183,6 +259,7 @@ int mxs_saif_get_mclk(unsigned int saif_id, unsigned int mclk, struct mxs_saif *saif = mxs_saif[saif_id]; u32 stat; int ret; + struct mxs_saif *master_saif; if (!saif) return -EINVAL; @@ -195,6 +272,12 @@ int mxs_saif_get_mclk(unsigned int saif_id, unsigned int mclk, __raw_writel(BM_SAIF_CTRL_CLKGATE, saif->base + SAIF_CTRL + MXS_CLR_ADDR); + master_saif = mxs_saif_get_master(saif); + if (saif != master_saif) { + dev_err(saif->dev, "can not get mclk from a non-master saif\n"); + return -EINVAL; + } + stat = __raw_readl(saif->base + SAIF_STAT); if (stat & BM_SAIF_STAT_BUSY) { dev_err(saif->dev, "error: busy\n"); @@ -278,10 +361,21 @@ static int mxs_saif_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) /* * Note: We simply just support master mode since SAIF TX can only * work as master. + * Here the master is relative to codec side. + * Saif internally could be slave when working on EXTMASTER mode. + * We just hide this to machine driver. */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: - scr &= ~BM_SAIF_CTRL_SLAVE_MODE; + if (saif->clkmux == MXS_DIGCTL_SAIF_CLKMUX_EXTMSTR0) + scr = saif->id ? scr | BM_SAIF_CTRL_SLAVE_MODE : + scr & ~BM_SAIF_CTRL_SLAVE_MODE; + else if (saif->clkmux == MXS_DIGCTL_SAIF_CLKMUX_EXTMSTR1) + scr = saif->id ? scr & ~BM_SAIF_CTRL_SLAVE_MODE : + scr | BM_SAIF_CTRL_SLAVE_MODE; + else + scr &= ~BM_SAIF_CTRL_SLAVE_MODE; + __raw_writel(scr | scr0, saif->base + SAIF_CTRL); break; default: @@ -396,6 +490,12 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai); + struct mxs_saif *master_saif; + u32 delay; + + master_saif = mxs_saif_get_master(saif); + if (!master_saif) + return -EINVAL; switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -403,10 +503,20 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: dev_dbg(cpu_dai->dev, "start\n"); - clk_enable(saif->clk); - if (!saif->mclk_in_use) + clk_enable(master_saif->clk); + if (!master_saif->mclk_in_use) + __raw_writel(BM_SAIF_CTRL_RUN, + master_saif->base + SAIF_CTRL + MXS_SET_ADDR); + + /* + * If the saif's master is not himself, we also need to enable + * itself clk for its internal basic logic to work. + */ + if (saif != master_saif) { + clk_enable(saif->clk); __raw_writel(BM_SAIF_CTRL_RUN, saif->base + SAIF_CTRL + MXS_SET_ADDR); + } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { /* @@ -422,20 +532,39 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, __raw_readl(saif->base + SAIF_DATA); } - dev_dbg(cpu_dai->dev, "CTRL 0x%x STAT 0x%x\n", + master_saif->ongoing = 1; + + dev_dbg(saif->dev, "CTRL 0x%x STAT 0x%x\n", __raw_readl(saif->base + SAIF_CTRL), __raw_readl(saif->base + SAIF_STAT)); + dev_dbg(master_saif->dev, "CTRL 0x%x STAT 0x%x\n", + __raw_readl(master_saif->base + SAIF_CTRL), + __raw_readl(master_saif->base + SAIF_STAT)); break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: dev_dbg(cpu_dai->dev, "stop\n"); - clk_disable(saif->clk); - if (!saif->mclk_in_use) + /* wait a while for the current sample to complete */ + delay = USEC_PER_SEC / master_saif->cur_rate; + + if (!master_saif->mclk_in_use) { + __raw_writel(BM_SAIF_CTRL_RUN, + master_saif->base + SAIF_CTRL + MXS_CLR_ADDR); + udelay(delay); + } + clk_disable(master_saif->clk); + + if (saif != master_saif) { __raw_writel(BM_SAIF_CTRL_RUN, saif->base + SAIF_CTRL + MXS_CLR_ADDR); + udelay(delay); + clk_disable(saif->clk); + } + + master_saif->ongoing = 0; break; default: @@ -519,15 +648,24 @@ static int mxs_saif_probe(struct platform_device *pdev) { struct resource *res; struct mxs_saif *saif; + struct mxs_saif_platform_data *pdata; int ret = 0; + pdata = pdev->dev.platform_data; + if (pdata && pdata->init()) + return -EINVAL; + saif = kzalloc(sizeof(*saif), GFP_KERNEL); if (!saif) return -ENOMEM; + if (pdata) + saif->clkmux = pdata->clkmux; + if (pdev->id >= ARRAY_SIZE(mxs_saif)) return -EINVAL; mxs_saif[pdev->id] = saif; + saif->id = pdev->id; saif->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(saif->clk)) { diff --git a/sound/soc/mxs/mxs-saif.h b/sound/soc/mxs/mxs-saif.h index 0e2ff8c..2a6aec1 100644 --- a/sound/soc/mxs/mxs-saif.h +++ b/sound/soc/mxs/mxs-saif.h @@ -118,6 +118,10 @@ struct mxs_saif { void __iomem *base; int irq; struct mxs_pcm_dma_params dma_param; + unsigned int id; + unsigned int clkmux; + unsigned int cur_rate; + unsigned int ongoing; struct platform_device *soc_platform_pdev; u32 fifo_underrun; -- 1.7.0.4

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Re: [alsa-devel] Throughts on getting ALC892 working better (was Re: Help With Getting Line-in monitoring working)
by Raymond Yau 06 Sep '11

06 Sep '11

2011/9/6 Kevin Atkinson <kevin(a)atkinson.dhs.org>: How did you get "snd: bad kfree" in the system log ? [ 6.272032] snd: bad kfree (called from f87381e3) -- [ 6.272043] [<f85612b1>] ? snd_hidden_kfree+0x91/0xa0 [snd] [ 6.272048] [<f87381e3>] ? snd_array_free+0x13/0x30 [snd_hda_codec] [ 6.272052] [<f87381e3>] ? snd_array_free+0x13/0x30 [snd_hda_codec] [ 6.272057] [<f87abfaf>] ? alc_free_kctls+0x4f/0x60 [snd_hda_codec_realtek] [ 6.272061] [<f87affed>] ? alc_build_controls+0x44d/0x4b0 [snd_hda_codec_realtek] > > I spent several hours going over the data sheet (ver 1.3, you can find it at > http://www.realtek.com/downloads/downloadsView.aspx?Langid=1&PNid=24&PFid=2… > > Apparently there is no hardware master control and the master and PCM slider > must be implemented in software. > > So, after the change you suggest below the mixer are mapped better in the > newer drivers. But things can be still be improved. More below. > > Also to get line-in pass-through working via headphone out please unmute the > second set of Amp-in vals of pin 0x26. > >> Node 0x26 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In >> Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >> Amp-In vals: [0x00 0x00] [0x80 0x80] > > ^^^^^^^^^^^ set these to 0x00 >> >> Connection: 2 >> 0x25 0x0b > The ALC892 provides ten DAC channels that simultaneously support 7.1 channel sound playback, plus 2 channels of independent stereo sound output (multiple streaming) through the front panel stereo outputs. you can create playback device 2 for the front panel headphone by adding spec->alt_dac_nid = 0x25; > Apparently all analog inputs are mixed together using 0x0B and then the > output is sent to a bunch of other audio mixers, but these mixers don't > really control the volume, they just have a separate mute for the PCM and > analog input. That being said, it probably also makes sense to mute analog > audio (coming from 0x0B) on all but pins 0x26 (headphone) and 0x0C (front) > by default. > The hardest thing is how to add the controls for this alc892 http://thread.gmane.org/gmane.linux.alsa.devel/88324 The laptop has built-in 5.1, and 5.1 or 7.1 output via jacks. DAC NIDs: 0x02 front 0x03 rear 0x04 clfe 0x05 side internal mixers: 0x0c front 0x0d rear 0x0e clfe 0x0f side internal speaker pins: 0x14 front 0x15 "rear" (front of laptop) 0x16 clfe external ports: fixed route: 0x17 side can select any output: 0x18 mic / output 0x1a line-in / output 0x1b headphone > > On an even lesser note, it would be really nice if ALSA could expose the > full capacity of the hardware without having an unwieldy number of mixer > controls. For example each of the 5 output mixers have a separate PCM and > analog pass-though mute control, it would be nice to expose this. Also, > each of the 8 analog pins can be repurposed as either input or output, some > of them have a headphone amp, some of them have a mic boost, it would be > nice if all this could all be exposed. However, I can see how exposing all > this will lead to a lot of confusing mixer settings. > does any user really want to pass the stereo signal from line in to "center/lfe" ?

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[alsa-devel] [PATCH] ASoC: Blackfin: bf5xx-ad193x: Fix codec device name
by Lars-Peter Clausen 06 Sep '11

06 Sep '11

Fix the codec_name field of the dai_link to match the actual device name of the codec. Otherwise the card won't be instantiated. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Cc: stable(a)kernel.org --- sound/soc/blackfin/bf5xx-ad193x.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ad193x.c b/sound/soc/blackfin/bf5xx-ad193x.c index a118a0f..5956584 100644 --- a/sound/soc/blackfin/bf5xx-ad193x.c +++ b/sound/soc/blackfin/bf5xx-ad193x.c @@ -103,7 +103,7 @@ static struct snd_soc_dai_link bf5xx_ad193x_dai[] = { .cpu_dai_name = "bfin-tdm.0", .codec_dai_name ="ad193x-hifi", .platform_name = "bfin-tdm-pcm-audio", - .codec_name = "ad193x.5", + .codec_name = "spi0.5", .ops = &bf5xx_ad193x_ops, }, { @@ -112,7 +112,7 @@ static struct snd_soc_dai_link bf5xx_ad193x_dai[] = { .cpu_dai_name = "bfin-tdm.1", .codec_dai_name ="ad193x-hifi", .platform_name = "bfin-tdm-pcm-audio", - .codec_name = "ad193x.5", + .codec_name = "spi0.5", .ops = &bf5xx_ad193x_ops, }, }; -- 1.7.2.5

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Re: [alsa-devel] [PATCH] ASoC: alc5623: Remove unused mutex
by Mark Brown 06 Sep '11

06 Sep '11

On Sun, Sep 04, 2011 at 10:48:27PM +0800, Axel Lin wrote: > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> > Cc: Arnaud Patard <arnaud.patard(a)rtp-net.org> Applied, thanks.

1 0

Re: [alsa-devel] [PATCH] ASoC: Add missing platform_device_put in raumfeld_audio_init error path
by Mark Brown 06 Sep '11

06 Sep '11

On Fri, Sep 02, 2011 at 09:47:41PM +0800, Axel Lin wrote: > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> Applied, thanks.

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[alsa-devel] [PATCH] ASoC: Check that WM8996 FLL started even if we don't have the IRQ
by Mark Brown 06 Sep '11

06 Sep '11

We can directly read the FLL lock status on WM8996 so even if we don't have an interrupt wired up we can still verify that the FLL started successfully. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 26 +++++++++++++++++++------- 1 files changed, 19 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 169a865..cbd0206 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2054,7 +2054,7 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source, struct i2c_client *i2c = to_i2c_client(codec->dev); struct _fll_div fll_div; unsigned long timeout; - int ret, reg; + int ret, reg, retry; /* Any change? */ if (source == wm8996->fll_src && Fref == wm8996->fll_fref && @@ -2141,17 +2141,29 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source, else timeout = msecs_to_jiffies(2); - /* Allow substantially longer if we've actually got the IRQ */ + /* Allow substantially longer if we've actually got the IRQ, poll + * at a slightly higher rate if we don't. + */ if (i2c->irq) - timeout *= 1000; + timeout *= 10; + else + timeout /= 2; - ret = wait_for_completion_timeout(&wm8996->fll_lock, timeout); + for (retry = 0; retry < 10; retry++) { + ret = wait_for_completion_timeout(&wm8996->fll_lock, + timeout); + if (ret != 0) { + WARN_ON(!i2c->irq); + break; + } - if (ret == 0 && i2c->irq) { + ret = snd_soc_read(codec, WM8996_INTERRUPT_RAW_STATUS_2); + if (ret & WM8996_FLL_LOCK_STS) + break; + } + if (retry == 10) { dev_err(codec->dev, "Timed out waiting for FLL\n"); ret = -ETIMEDOUT; - } else { - ret = 0; } dev_dbg(codec->dev, "FLL configured for %dHz->%dHz\n", Fref, Fout); -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: Display the error code when we fail to add a DAPM control
by Mark Brown 06 Sep '11

06 Sep '11

Useful for diagnostics. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 170c4ff..681afc5 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -584,8 +584,8 @@ static int dapm_new_mux(struct snd_soc_dapm_widget *w) name + prefix_len, prefix); ret = snd_ctl_add(card, kcontrol); if (ret < 0) { - dev_err(dapm->dev, - "asoc: failed to add kcontrol %s\n", w->name); + dev_err(dapm->dev, "failed to add kcontrol %s: %d\n", + w->name, ret); kfree(wlist); return ret; } -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: Check that WM8996 FLL started even if we don't have the IRQ
by Mark Brown 06 Sep '11

06 Sep '11

We can directly read the FLL lock status on WM8996 so even if we don't have an interrupt wired up we can still verify that the FLL started successfully. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 26 +++++++++++++++++++------- 1 files changed, 19 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 169a865..cbd0206 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2054,7 +2054,7 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source, struct i2c_client *i2c = to_i2c_client(codec->dev); struct _fll_div fll_div; unsigned long timeout; - int ret, reg; + int ret, reg, retry; /* Any change? */ if (source == wm8996->fll_src && Fref == wm8996->fll_fref && @@ -2141,17 +2141,29 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source, else timeout = msecs_to_jiffies(2); - /* Allow substantially longer if we've actually got the IRQ */ + /* Allow substantially longer if we've actually got the IRQ, poll + * at a slightly higher rate if we don't. + */ if (i2c->irq) - timeout *= 1000; + timeout *= 10; + else + timeout /= 2; - ret = wait_for_completion_timeout(&wm8996->fll_lock, timeout); + for (retry = 0; retry < 10; retry++) { + ret = wait_for_completion_timeout(&wm8996->fll_lock, + timeout); + if (ret != 0) { + WARN_ON(!i2c->irq); + break; + } - if (ret == 0 && i2c->irq) { + ret = snd_soc_read(codec, WM8996_INTERRUPT_RAW_STATUS_2); + if (ret & WM8996_FLL_LOCK_STS) + break; + } + if (retry == 10) { dev_err(codec->dev, "Timed out waiting for FLL\n"); ret = -ETIMEDOUT; - } else { - ret = 0; } dev_dbg(codec->dev, "FLL configured for %dHz->%dHz\n", Fref, Fout); -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: Fix reporting of partial jack updates
by Mark Brown 06 Sep '11

06 Sep '11

We need to report the entire jack state to the core jack code, not just the bits that were being updated by the caller, otherwise the status reported by other detection methods will be omitted from the state seen by userspace. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-jack.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 38b0013..fa31d9c 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -105,7 +105,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) snd_soc_dapm_sync(dapm); - snd_jack_report(jack->jack, status); + snd_jack_report(jack->jack, jack->status); out: mutex_unlock(&codec->mutex); -- 1.7.5.4

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