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[alsa-devel] [PATCH] [PATCH]ASoC:blackfin:add ad73311 card driver name
by Scott Jiang 07 Sep '11

07 Sep '11

ASoC doesn't copy codec name to card driver now. ALSA lib need this info to find the proper card config. So add card name in snd_soc_card, asoc will copy this name to snd_card. Signed-off-by: Scott Jiang <scott.jiang.linux(a)gmail.com> --- sound/soc/blackfin/bf5xx-ad73311.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c index 732a247..975c6e8 100644 --- a/sound/soc/blackfin/bf5xx-ad73311.c +++ b/sound/soc/blackfin/bf5xx-ad73311.c @@ -192,6 +192,7 @@ static struct snd_soc_dai_link bf5xx_ad73311_dai[] = { static struct snd_soc_card bf5xx_ad73311 = { .name = "bfin-ad73311", + .driver_name = "AD73311", .probe = bf5xx_probe, .dai_link = &bf5xx_ad73311_dai[CONFIG_SND_BF5XX_SPORT_NUM], .num_links = 1, -- 1.7.0.4

2 2

Re: [alsa-devel] [PATCH] ASoC: Add driver for rt5631
by Mark Brown 06 Sep '11

06 Sep '11

On Tue, Sep 06, 2011 at 06:32:42PM +0800, johnnyhsu(a)realtek.com wrote: > switch (reg) { > - case RT5631_VENDOR_ID: > - case RT5631_VENDOR_ID1: > - case RT5631_VENDOR_ID2: > - return 0; > - default: > + case RT5631_RESET: > + case RT5631_INT_ST_IRQ_CTRL_2: > + case RT5631_INDEX_ADD: > + case RT5631_INDEX_DATA: > + case RT5631_EQ_CTRL: > return 1; > + default: > + return 0; This is an incremental patch against your previous driver, not a full patch. You should always send a patches which can be applied against current kernels. > - pr_info("enter %s, syclk=%d\n", __func__, freq); > + dev_info(codec->dev, "enter %s, syclk=%d\n", __func__, freq); This should be a debug level print at most, otherwise you just spam the console. > if (!freq_in || !freq_out) { > dev_dbg(codec->dev, "PLL disabled\n"); > - return -EINVAL; > + return 0; I'd expect to see some sort of register write when the PLL is disabled? > /** > - * rt5631_index_reg_show - sysfs file for dumping index registers of 2nd layer > + * rt5631_index_reg_show: Show private register of rt5631 codec. > + * @dev: device to query. > + * @attr: device attribute. > + * @buf: buffer to display. > + * > + * To show private registers which are not changed often by user. > + * You have to access them through register 0x6a and 0x6c. > */ > static ssize_t rt5631_index_reg_show(struct device *dev, > struct device_attribute *attr, char *buf) As I've told you repeatedly you should be using the standard facilities for debug access to registers - you should not open coding this in your driver. Once again, ignoring review comments isn't helpful.

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[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

3 4

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" ?

2 1

[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.

2 2

[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

2 1

[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

2 1

[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

2 1

[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

2 1

[alsa-devel] [PATCH v2 1/2] regmap: Add support for device specific write and read flag masks.
by Lars-Peter Clausen 06 Sep '11

06 Sep '11

Some buses like SPI have no standard notation of read or write operations. The general scheme here is to set or clear specific bits in the register address to indicate whether the operation is a read or write. We already support having a read flag mask per bus, but as there is no standard the bits which need to be set or cleared differ between devices and vendors, thus we need a mechanism to specify them per device. This patch adds two new entries to the regmap_config struct, read_flag_mask and write_flag_mask. These will be or'ed onto the top byte when doing a read or write operation. If both masks are empty the device will fallback to the regmap_bus masks. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- No changes since v1 --- drivers/base/regmap/internal.h | 3 +++ drivers/base/regmap/regmap.c | 15 ++++++++++++--- include/linux/regmap.h | 9 +++++++++ 3 files changed, 24 insertions(+), 3 deletions(-) diff --git a/drivers/base/regmap/internal.h b/drivers/base/regmap/internal.h index 5ab3fef..7e14d5a 100644 --- a/drivers/base/regmap/internal.h +++ b/drivers/base/regmap/internal.h @@ -46,6 +46,9 @@ struct regmap { bool (*readable_reg)(struct device *dev, unsigned int reg); bool (*volatile_reg)(struct device *dev, unsigned int reg); bool (*precious_reg)(struct device *dev, unsigned int reg); + + u8 read_flag_mask; + u8 write_flag_mask; }; bool regmap_writeable(struct regmap *map, unsigned int reg); diff --git a/drivers/base/regmap/regmap.c b/drivers/base/regmap/regmap.c index fa2bd89..8ecf8bf 100644 --- a/drivers/base/regmap/regmap.c +++ b/drivers/base/regmap/regmap.c @@ -147,6 +147,13 @@ struct regmap *regmap_init(struct device *dev, map->volatile_reg = config->volatile_reg; map->precious_reg = config->precious_reg; + if (config->read_flag_mask || config->write_flag_mask) { + map->read_flag_mask = config->read_flag_mask; + map->write_flag_mask = config->write_flag_mask; + } else { + map->read_flag_mask = bus->read_flag_mask; + } + switch (config->reg_bits) { case 4: switch (config->val_bits) { @@ -229,6 +236,7 @@ EXPORT_SYMBOL_GPL(regmap_exit); static int _regmap_raw_write(struct regmap *map, unsigned int reg, const void *val, size_t val_len) { + u8 *u8 = map->work_buf; void *buf; int ret = -ENOTSUPP; size_t len; @@ -242,6 +250,8 @@ static int _regmap_raw_write(struct regmap *map, unsigned int reg, map->format.format_reg(map->work_buf, reg); + u8[0] |= map->write_flag_mask; + trace_regmap_hw_write_start(map->dev, reg, val_len / map->format.val_bytes); @@ -369,13 +379,12 @@ static int _regmap_raw_read(struct regmap *map, unsigned int reg, void *val, map->format.format_reg(map->work_buf, reg); /* - * Some buses flag reads by setting the high bits in the + * Some buses or devices flag reads by setting the high bits in the * register addresss; since it's always the high bits for all * current formats we can do this here rather than in * formatting. This may break if we get interesting formats. */ - if (map->bus->read_flag_mask) - u8[0] |= map->bus->read_flag_mask; + u8[0] |= map->read_flag_mask; trace_regmap_hw_read_start(map->dev, reg, val_len / map->format.val_bytes); diff --git a/include/linux/regmap.h b/include/linux/regmap.h index 449e264..f0b8d47 100644 --- a/include/linux/regmap.h +++ b/include/linux/regmap.h @@ -53,6 +53,12 @@ struct reg_default { * @reg_defaults: Power on reset values for registers (for use with * register cache support). * @num_reg_defaults: Number of elements in reg_defaults. + * + * @read_flag_mask: Mask to be set in the top byte of the register when doing + * a read. + * @write_flag_mask: Mask to be set in the top byte of the register when doing + * a write. If both read_flag_mask and write_flag_mask are + * empty the regmap_bus default masks are used. */ struct regmap_config { int reg_bits; @@ -66,6 +72,9 @@ struct regmap_config { unsigned int max_register; struct reg_default *reg_defaults; int num_reg_defaults; + + u8 read_flag_mask; + u8 write_flag_mask; }; typedef int (*regmap_hw_write)(struct device *dev, const void *data, -- 1.7.2.5

3 3

Re: [alsa-devel] [PATCH] ASoC: alc5623: Remove unused mutex
by Girdwood, Liam 06 Sep '11

06 Sep '11

On 4 September 2011 15:48, Axel Lin <axel.lin(a)gmail.com> wrote: > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> > Cc: Arnaud Patard <arnaud.patard(a)rtp-net.org> > --- > sound/soc/codecs/alc5623.c | 2 -- > 1 files changed, 0 insertions(+), 2 deletions(-) > > diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c > index eecffb5..0517315 100644 > --- a/sound/soc/codecs/alc5623.c > +++ b/sound/soc/codecs/alc5623.c > @@ -41,7 +41,6 @@ MODULE_PARM_DESC(caps_charge, "ALC5623 cap charge time > (msecs)"); > struct alc5623_priv { > enum snd_soc_control_type control_type; > void *control_data; > - struct mutex mutex; > u8 id; > unsigned int sysclk; > u16 reg_cache[ALC5623_VENDOR_ID2+2]; > @@ -1052,7 +1051,6 @@ static int alc5623_i2c_probe(struct i2c_client > *client, > i2c_set_clientdata(client, alc5623); > alc5623->control_data = client; > alc5623->control_type = SND_SOC_I2C; > - mutex_init(&alc5623->mutex); > > ret = snd_soc_register_codec(&client->dev, > &soc_codec_device_alc5623, &alc5623_dai, 1); > -- > 1.7.4.1 > > > > Acked-by: Liam Girdwood <lrg(a)ti.com>

1 0

[alsa-devel] Help With Getting Line-in monitoring working
by Kevin Atkinson 06 Sep '11

06 Sep '11

Hi, I need some help getting my cards line-in monitoring working. That is analog pass-through from line-in to headphone out. I am using Ubuntu 10.04 with Linux kernel Linux Intel 2.6.38-10 (Ubuntu package). I am using on-board audio with HDA-Intel with the Realtek ALC892 codec. With the stock Ubuntu kernel (lets call this setup "stable") I can't get line-in monitoring working (note however I can record from line in, I just can't get the analog pass-through working). However, when I install the latest ALSA drivers (lets call this setup "soso") (recompiled a source package from the Ubuntu-audio-dev ppa, head 7c48f32b). I can get Line-in monitoring but the mixer controls are all mixed up. The Line-in control controls the Line-in volume, but the master control has no effect. Also, the PCM control has no effect on PCM volume and the front control controls the output to the headphone but in order to mute the headphone I still need to use the headphone mute. The mixer work correctly with the stable setup. Thus I am lead to the conclusion that the latest ALSA drivers are doing something right with line-in but getting a lot of other things wrong. Attached is the alsa-info output of both setups. Please give me some guidance on how to debug this problem. For example what is the newer code doing that the old code is not that gets line-in monitoring working but gets a lot of the other mixer setups wrong. I assume it is just a matter of "hooking up" the line in to the headphone out, and all that is needed is the right verb, but I am at a complete lost to what verb that is. If there is some specific version of the ALSA drivers I should try please let me know. If the Ubuntu kernel is causing a problem I am willing to install a stock kernel, but I rather not unless absolutely necessary.

2 2

[alsa-devel] [PATCH 1/3] regmap: Add support for device specific write and read flag masks.
by Lars-Peter Clausen 05 Sep '11

05 Sep '11

Some buses like SPI have no standard notation of read or write operations. The general scheme here is to set or clear specific bits in the register address to indicate whether the operation is a read or write. We already support having a read flag mask per bus, but as there is no standard the bits which need to be set or cleared differ between devices and vendors, thus we need a mechanism to specify them per device. This patch adds two new entries to the regmap_config struct, read_flag_mask and write_flag_mask. These will be or'ed onto the top byte when doing a read or write operation. If both masks are empty the device will fallback to the regmap_bus masks. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- drivers/base/regmap/internal.h | 3 +++ drivers/base/regmap/regmap.c | 15 ++++++++++++--- include/linux/regmap.h | 9 +++++++++ 3 files changed, 24 insertions(+), 3 deletions(-) diff --git a/drivers/base/regmap/internal.h b/drivers/base/regmap/internal.h index 5ab3fef..7e14d5a 100644 --- a/drivers/base/regmap/internal.h +++ b/drivers/base/regmap/internal.h @@ -46,6 +46,9 @@ struct regmap { bool (*readable_reg)(struct device *dev, unsigned int reg); bool (*volatile_reg)(struct device *dev, unsigned int reg); bool (*precious_reg)(struct device *dev, unsigned int reg); + + u8 read_flag_mask; + u8 write_flag_mask; }; bool regmap_writeable(struct regmap *map, unsigned int reg); diff --git a/drivers/base/regmap/regmap.c b/drivers/base/regmap/regmap.c index fa2bd89..8ecf8bf 100644 --- a/drivers/base/regmap/regmap.c +++ b/drivers/base/regmap/regmap.c @@ -147,6 +147,13 @@ struct regmap *regmap_init(struct device *dev, map->volatile_reg = config->volatile_reg; map->precious_reg = config->precious_reg; + if (config->read_flag_mask || config->write_flag_mask) { + map->read_flag_mask = config->read_flag_mask; + map->write_flag_mask = config->write_flag_mask; + } else { + map->read_flag_mask = bus->read_flag_mask; + } + switch (config->reg_bits) { case 4: switch (config->val_bits) { @@ -229,6 +236,7 @@ EXPORT_SYMBOL_GPL(regmap_exit); static int _regmap_raw_write(struct regmap *map, unsigned int reg, const void *val, size_t val_len) { + u8 *u8 = map->work_buf; void *buf; int ret = -ENOTSUPP; size_t len; @@ -242,6 +250,8 @@ static int _regmap_raw_write(struct regmap *map, unsigned int reg, map->format.format_reg(map->work_buf, reg); + u8[0] |= map->write_flag_mask; + trace_regmap_hw_write_start(map->dev, reg, val_len / map->format.val_bytes); @@ -369,13 +379,12 @@ static int _regmap_raw_read(struct regmap *map, unsigned int reg, void *val, map->format.format_reg(map->work_buf, reg); /* - * Some buses flag reads by setting the high bits in the + * Some buses or devices flag reads by setting the high bits in the * register addresss; since it's always the high bits for all * current formats we can do this here rather than in * formatting. This may break if we get interesting formats. */ - if (map->bus->read_flag_mask) - u8[0] |= map->bus->read_flag_mask; + u8[0] |= map->read_flag_mask; trace_regmap_hw_read_start(map->dev, reg, val_len / map->format.val_bytes); diff --git a/include/linux/regmap.h b/include/linux/regmap.h index 449e264..f0b8d47 100644 --- a/include/linux/regmap.h +++ b/include/linux/regmap.h @@ -53,6 +53,12 @@ struct reg_default { * @reg_defaults: Power on reset values for registers (for use with * register cache support). * @num_reg_defaults: Number of elements in reg_defaults. + * + * @read_flag_mask: Mask to be set in the top byte of the register when doing + * a read. + * @write_flag_mask: Mask to be set in the top byte of the register when doing + * a write. If both read_flag_mask and write_flag_mask are + * empty the regmap_bus default masks are used. */ struct regmap_config { int reg_bits; @@ -66,6 +72,9 @@ struct regmap_config { unsigned int max_register; struct reg_default *reg_defaults; int num_reg_defaults; + + u8 read_flag_mask; + u8 write_flag_mask; }; typedef int (*regmap_hw_write)(struct device *dev, const void *data, -- 1.7.2.5

2 5

Re: [alsa-devel] [PATCH v5] drm/i915: pass ELD to HDMI/DP audio driver
by Wu Fengguang 05 Sep '11

05 Sep '11

On Mon, Sep 05, 2011 at 07:04:50PM +0800, Chris Wilson wrote: > On Mon, 5 Sep 2011 09:14:00 +0800, Wu Fengguang <fengguang.wu(a)intel.com> wrote: > > On Sun, Sep 04, 2011 at 08:08:37PM +0800, Chris Wilson wrote: > > > On Sun, 4 Sep 2011 05:15:10 +0800, Wu Fengguang <fengguang.wu(a)intel.com> wrote: > > > This patch should be split between adding the core drm functionality to > > > build the ELD and the introduction of i915 support. > > > > OK. I didn't do this because I was not sure if it's OK to just add the > > drm_*() functions without any code to call them.. > > Right, we don't introduce new interfaces without users - but it is > acceptable to say "... which will be used by i915 in a following patch" if > you want to clarify the need for the interface. It just helps to > compartmentalize the damage should we find something goes horribly wrong > later. Sounds pretty reasonable. And the logical split may help make a bit easier to review :) Fengguang

1 0

[alsa-devel] [PATCH] snd_mixer_poll_descriptors_revents: iterate over all descriptors
by Olivier Trichet 05 Sep '11

05 Sep '11

Hello, Reading the method snd_mixer_poll_descriptors_revents, it seems to me that the iteration over the array 'pfds' is wrong. Shouldn't the attached patch be applied? Regards, Olivier

2 2

Re: [alsa-devel] [PATCH] ASoC/MFD: twl4030: Fix dependencies for audio codec
by Péter Ujfalusi 05 Sep '11

05 Sep '11

On Monday 05 September 2011 11:26:33 Thomas Weber wrote: > The codec for Devkit8000 (TWL4030) was not detected except > when build with CONFIG_SND_SOC_ALL_CODECS. > > twl-core.c still uses the CONFIG_TWL4030_CODEC for > twl_has_codec(). acked-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com>

1 0

[alsa-devel] [PATCH] ALSA: usb: refine delay information with USB frame counter
by Pierre-Louis Bossart 05 Sep '11

05 Sep '11

Existing code only updates the audio delay when URBs were submitted/retired. This can introduce an uncertainty of 8ms on the number of samples played out with the default settings, and a lot more when URBs convey more packets to reduce the interrupt rate and power consumption. This patch relies on the USB frame counter to reduce the uncertainty to less than 2ms worst-case. The delay information essentially becomes independent of the URB size and number of packets. This should improve audio/video sync and help applications like PulseAudio which require accurate audio timing. As pointed out by Clemens Ladisch <clemens(a)ladisch.de>, the implementation of frame counters varies between different HCDs. Only the 8 lowest-bits are used to estimate the delay. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/usb/card.h | 2 ++ sound/usb/pcm.c | 30 ++++++++++++++++++++++++++++++ sound/usb/pcm.h | 3 +++ sound/usb/urb.c | 28 +++++++++++++++++++++++++--- 4 files changed, 60 insertions(+), 3 deletions(-) diff --git a/sound/usb/card.h b/sound/usb/card.h index ae4251d..a39edcc 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -94,6 +94,8 @@ struct snd_usb_substream { spinlock_t lock; struct snd_urb_ops ops; /* callbacks (must be filled at init) */ + int last_frame_number; /* stored frame number */ + int last_delay; /* stored delay */ }; struct snd_usb_stream { diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index b8dcbf4..6076863 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -34,6 +34,32 @@ #include "clock.h" #include "power.h" +/* return the estimated delay based on USB frame counters */ +snd_pcm_uframes_t snd_usb_pcm_delay(struct snd_usb_substream *subs, + unsigned int rate) +{ + int current_frame_number; + int frame_diff; + int est_delay; + + current_frame_number = usb_get_current_frame_number(subs->dev); + /* HCD implementations use different widths, use lower 8 bits. + * The delay will be managed up to 256ms, which is more than + * enough + */ + current_frame_number &= 0xFF; + frame_diff = current_frame_number-subs->last_frame_number; + if (frame_diff < 0) + frame_diff += 256; /* handle 8-bit wrap-around */ + + /* Approximation based on number of samples per USB frame (ms), + some truncation for 44.1 but the estimate is good enough */ + est_delay = subs->last_delay-(frame_diff*rate/1000L); + if (est_delay < 0) + est_delay = 0; + return est_delay; +} + /* * return the current pcm pointer. just based on the hwptr_done value. */ @@ -45,6 +71,8 @@ static snd_pcm_uframes_t snd_usb_pcm_pointer(struct snd_pcm_substream *substream subs = (struct snd_usb_substream *)substream->runtime->private_data; spin_lock(&subs->lock); hwptr_done = subs->hwptr_done; + substream->runtime->delay = snd_usb_pcm_delay(subs, + substream->runtime->rate); spin_unlock(&subs->lock); return hwptr_done / (substream->runtime->frame_bits >> 3); } @@ -417,6 +445,8 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) subs->hwptr_done = 0; subs->transfer_done = 0; subs->phase = 0; + subs->last_delay = 0; + subs->last_frame_number = 0; runtime->delay = 0; return snd_usb_substream_prepare(subs, runtime); diff --git a/sound/usb/pcm.h b/sound/usb/pcm.h index ed3e283..df7a003 100644 --- a/sound/usb/pcm.h +++ b/sound/usb/pcm.h @@ -1,6 +1,9 @@ #ifndef __USBAUDIO_PCM_H #define __USBAUDIO_PCM_H +snd_pcm_uframes_t snd_usb_pcm_delay(struct snd_usb_substream *subs, + unsigned int rate); + void snd_usb_set_pcm_ops(struct snd_pcm *pcm, int stream); int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface, diff --git a/sound/usb/urb.c b/sound/usb/urb.c index e184349..9e3e3f9 100644 --- a/sound/usb/urb.c +++ b/sound/usb/urb.c @@ -718,7 +718,16 @@ static int prepare_playback_urb(struct snd_usb_substream *subs, subs->hwptr_done += bytes; if (subs->hwptr_done >= runtime->buffer_size * stride) subs->hwptr_done -= runtime->buffer_size * stride; + + /* update delay with exact number of samples queued */ + runtime->delay = subs->last_delay; runtime->delay += frames; + subs->last_delay = runtime->delay; + + /* realign last_frame_number */ + subs->last_frame_number = usb_get_current_frame_number(subs->dev); + subs->last_frame_number &= 0xFF; /* keep 8 LSBs */ + spin_unlock_irqrestore(&subs->lock, flags); urb->transfer_buffer_length = bytes; if (period_elapsed) @@ -737,12 +746,25 @@ static int retire_playback_urb(struct snd_usb_substream *subs, unsigned long flags; int stride = runtime->frame_bits >> 3; int processed = urb->transfer_buffer_length / stride; + int est_delay; spin_lock_irqsave(&subs->lock, flags); - if (processed > runtime->delay) - runtime->delay = 0; + + est_delay = snd_usb_pcm_delay(subs, runtime->rate); + /* update delay with exact number of samples played */ + if (processed > subs->last_delay) + subs->last_delay = 0; else - runtime->delay -= processed; + subs->last_delay -= processed; + runtime->delay = subs->last_delay; + + /* Report when delay estimate is off by more than 2ms. + * The error should be lower than 2ms since the estimate relies + * on two reads of a counter updated every ms */ + if (abs(est_delay-subs->last_delay) > (runtime->rate*2/1000L)) + snd_printk(KERN_DEBUG "ALSA usb.c: Delay %d actual delay %d\n", + est_delay, subs->last_delay); + spin_unlock_irqrestore(&subs->lock, flags); return 0; } -- 1.7.6

2 3

[alsa-devel] [PATCH 1/2] drm: support routines for HDMI/DP ELD
by Wu Fengguang 05 Sep '11

05 Sep '11

ELD (EDID-Like Data) describes to the HDMI/DP audio driver the audio capabilities of the plugged monitor. This adds drm_edid_to_eld() for converting EDID to ELD. The converted ELD will be saved in a new drm_connector.eld[128] data field. This is necessary because the graphics driver will need to fixup some of the data fields (eg. HDMI/DP connection type, AV sync delay) before writing to the hardware ELD buffer. drm_av_sync_delay() will help the graphics drivers dynamically compute the AV sync delay for fixing-up the ELD. ELD selection policy: it's possible for one encoder to be associated with multiple connectors (ie. monitors), in which case the first found ELD will be returned by drm_select_eld(). This policy may not be suitable for all users, but let's start it simple first. The impact of ELD selection policy: assume there are two monitors, one supports stereo playback and the other has 8-channel output; cloned display mode is used, so that the two monitors are associated with the same internal encoder. If only the stereo playback capability is reported, the user won't be able to start 8-channel playback; if the 8-channel ELD is reported, then user space applications may send 8-channel samples down, however the user may actually be listening to the 2-channel monitor and not connecting speakers to the 8-channel monitor. According to James, many TVs will either refuse the display anything or pop-up an OSD warning whenever they receive hdmi audio which they cannot handle. Eventually we will require configurability and/or per-monitor audio control even when the video is cloned. CC: Zhao Yakui <yakui.zhao(a)intel.com> CC: Wang Zhenyu <zhenyu.z.wang(a)intel.com> CC: Jeremy Bush <contractfrombelow(a)gmail.com> CC: Christopher White <c.white(a)pulseforce.com> CC: Pierre-Louis Bossart <pierre-louis.bossart(a)intel.com> CC: Paul Menzel <paulepanter(a)users.sourceforge.net> CC: James Cloos <cloos(a)jhcloos.com> CC: Chris Wilson <chris(a)chris-wilson.co.uk> Signed-off-by: Ben Skeggs <bskeggs(a)redhat.com> Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- drivers/gpu/drm/drm_edid.c | 171 +++++++++++++++++++++++++++++++++++ include/drm/drm_edid.h | 9 + 2 files changed, 180 insertions(+) --- linux-next.orig/include/drm/drm_crtc.h 2011-09-04 09:42:19.000000000 +0800 +++ linux-next/include/drm/drm_crtc.h 2011-09-05 09:00:25.000000000 +0800 @@ -466,6 +466,8 @@ enum drm_connector_force { /* DACs should rarely do this without a lot of testing */ #define DRM_CONNECTOR_POLL_DISCONNECT (1 << 2) +#define MAX_ELD_BYTES 128 + /** * drm_connector - central DRM connector control structure * @crtc: CRTC this connector is currently connected to, NULL if none @@ -523,6 +525,13 @@ struct drm_connector { uint32_t force_encoder_id; struct drm_encoder *encoder; /* currently active encoder */ + /* EDID bits */ + uint8_t eld[MAX_ELD_BYTES]; + bool dvi_dual; + int max_tmds_clock; /* in MHz */ + bool latency_present[2]; + int video_latency[2]; /* [0]: progressive, [1]: interlaced */ + int audio_latency[2]; int null_edid_counter; /* needed to workaround some HW bugs where we get all 0s */ }; --- linux-next.orig/include/drm/drm_edid.h 2011-09-05 12:35:36.000000000 +0800 +++ linux-next/include/drm/drm_edid.h 2011-09-05 12:36:10.000000000 +0800 @@ -230,4 +230,13 @@ struct edid { #define EDID_PRODUCT_ID(e) ((e)->prod_code[0] | ((e)->prod_code[1] << 8)) +struct drm_encoder; +struct drm_connector; +struct drm_display_mode; +void drm_edid_to_eld(struct drm_connector *connector, struct edid *edid); +int drm_av_sync_delay(struct drm_connector *connector, + struct drm_display_mode *mode); +struct drm_connector *drm_select_eld(struct drm_encoder *encoder, + struct drm_display_mode *mode); + #endif /* __DRM_EDID_H__ */ --- linux-next.orig/drivers/gpu/drm/drm_edid.c 2011-09-05 12:35:36.000000000 +0800 +++ linux-next/drivers/gpu/drm/drm_edid.c 2011-09-05 12:36:10.000000000 +0800 @@ -1319,6 +1319,7 @@ add_detailed_modes(struct drm_connector #define HDMI_IDENTIFIER 0x000C03 #define AUDIO_BLOCK 0x01 #define VENDOR_BLOCK 0x03 +#define SPEAKER_BLOCK 0x04 #define EDID_BASIC_AUDIO (1 << 6) /** @@ -1347,6 +1348,176 @@ u8 *drm_find_cea_extension(struct edid * } EXPORT_SYMBOL(drm_find_cea_extension); +static void +parse_hdmi_vsdb(struct drm_connector *connector, uint8_t *db) +{ + connector->eld[5] |= (db[6] >> 7) << 1; /* Supports_AI */ + + connector->dvi_dual = db[6] & 1; + connector->max_tmds_clock = db[7] * 5; + + connector->latency_present[0] = db[8] >> 7; + connector->latency_present[1] = (db[8] >> 6) & 1; + connector->video_latency[0] = db[9]; + connector->audio_latency[0] = db[10]; + connector->video_latency[1] = db[11]; + connector->audio_latency[1] = db[12]; + + DRM_LOG_KMS("HDMI: DVI dual %d, " + "max TMDS clock %d, " + "latency present %d %d, " + "video latency %d %d, " + "audio latency %d %d\n", + connector->dvi_dual, + connector->max_tmds_clock, + (int) connector->latency_present[0], + (int) connector->latency_present[1], + connector->video_latency[0], + connector->video_latency[1], + connector->audio_latency[0], + connector->audio_latency[1]); +} + +static void +monitor_name(struct detailed_timing *t, void *data) +{ + if (t->data.other_data.type == EDID_DETAIL_MONITOR_NAME) + *(u8 **)data = t->data.other_data.data.str.str; +} + +/** + * drm_edid_to_eld - build ELD from EDID + * @connector: connector corresponding to the HDMI/DP sink + * @edid: EDID to parse + * + * Fill the ELD (EDID-Like Data) buffer for passing to the audio driver. + * Some ELD fields are left to the graphics driver caller: + * - Conn_Type + * - HDCP + * - Port_ID + */ +void drm_edid_to_eld(struct drm_connector *connector, struct edid *edid) +{ + uint8_t *eld = connector->eld; + u8 *cea; + u8 *name; + u8 *db; + int sad_count = 0; + int mnl; + int dbl; + + memset(eld, 0, sizeof(connector->eld)); + + cea = drm_find_cea_extension(edid); + if (!cea) { + DRM_DEBUG_KMS("ELD: no CEA Extension found\n"); + return; + } + + name = NULL; + drm_for_each_detailed_block((u8 *)edid, monitor_name, &name); + for (mnl = 0; name && mnl < 13; mnl++) { + if (name[mnl] == 0x0a) + break; + eld[20 + mnl] = name[mnl]; + } + eld[4] = (cea[1] << 5) | mnl; + DRM_DEBUG_KMS("ELD monitor %s\n", eld + 20); + + eld[0] = 2 << 3; /* ELD version: 2 */ + + eld[16] = edid->mfg_id[0]; + eld[17] = edid->mfg_id[1]; + eld[18] = edid->prod_code[0]; + eld[19] = edid->prod_code[1]; + + for (db = cea + 4; db < cea + cea[2]; db += dbl + 1) { + dbl = db[0] & 0x1f; + + switch ((db[0] & 0xe0) >> 5) { + case AUDIO_BLOCK: /* Audio Data Block, contains SADs */ + sad_count = dbl / 3; + memcpy(eld + 20 + mnl, &db[1], dbl); + break; + case SPEAKER_BLOCK: /* Speaker Allocation Data Block */ + eld[7] = db[1]; + break; + case VENDOR_BLOCK: + /* HDMI Vendor-Specific Data Block */ + if (db[1] == 0x03 && db[2] == 0x0c && db[3] == 0) + parse_hdmi_vsdb(connector, db); + break; + default: + break; + } + } + eld[5] |= sad_count << 4; + eld[2] = (20 + mnl + sad_count * 3 + 3) / 4; + + DRM_DEBUG_KMS("ELD size %d, SAD count %d\n", (int)eld[2], sad_count); +} +EXPORT_SYMBOL(drm_edid_to_eld); + +/** + * drm_av_sync_delay - HDMI/DP sink audio-video sync delay in millisecond + * @connector: connector associated with the HDMI/DP sink + * @mode: the display mode + */ +int drm_av_sync_delay(struct drm_connector *connector, + struct drm_display_mode *mode) +{ + int i = !!(mode->flags & DRM_MODE_FLAG_INTERLACE); + int a, v; + + if (!connector->latency_present[0]) + return 0; + if (!connector->latency_present[1]) + i = 0; + + a = connector->audio_latency[i]; + v = connector->video_latency[i]; + + /* + * HDMI/DP sink doesn't support audio or video? + */ + if (a == 255 || v == 255) + return 0; + + /* + * Convert raw EDID values to millisecond. + * Treat unknown latency as 0ms. + */ + if (a) + a = min(2 * (a - 1), 500); + if (v) + v = min(2 * (v - 1), 500); + + return max(v - a, 0); +} +EXPORT_SYMBOL(drm_av_sync_delay); + +/** + * drm_select_eld - select one ELD from multiple HDMI/DP sinks + * @encoder: the encoder just changed display mode + * @mode: the adjusted display mode + * + * It's possible for one encoder to be associated with multiple HDMI/DP sinks. + * The policy is now hard coded to simply use the first HDMI/DP sink's ELD. + */ +struct drm_connector *drm_select_eld(struct drm_encoder *encoder, + struct drm_display_mode *mode) +{ + struct drm_connector *connector; + struct drm_device *dev = encoder->dev; + + list_for_each_entry(connector, &dev->mode_config.connector_list, head) + if (connector->encoder == encoder && connector->eld[0]) + return connector; + + return NULL; +} +EXPORT_SYMBOL(drm_select_eld); + /** * drm_detect_hdmi_monitor - detect whether monitor is hdmi. * @edid: monitor EDID information

1 3

[alsa-devel] [PATCH v4] drm/i915: pass ELD to HDMI/DP audio driver
by Wu Fengguang 05 Sep '11

05 Sep '11

Add ELD support for Intel Eaglelake, IbexPeak/Ironlake, SandyBridge/CougarPoint and IvyBridge/PantherPoint chips. ELD (EDID-Like Data) describes to the HDMI/DP audio driver the audio capabilities of the plugged monitor. It's built and passed to audio driver in 2 steps: (1) at get_modes time, parse EDID and save ELD to drm_connector.eld[] (2) at mode_set time, write drm_connector.eld[] to the Transcoder's hw ELD buffer and set the ELD_valid bit to inform HDMI/DP audio driver ELD selection policy: it's possible for one encoder to be associated with multiple connectors (ie. monitors), in which case the first found ELD will be used. This policy may not be suitable for all users, but let's start it simple first. The impact of ELD selection policy: assume there are two monitors, one supports stereo playback and the other has 8-channel output; cloned display mode is used, so that the two monitors are associated with the same internal encoder. If only the stereo playback capability is reported, the user won't be able to start 8-channel playback; if the 8-channel ELD is reported, then user space applications may send 8-channel samples down, however the user may actually be listening to the 2-channel monitor and not connecting speakers to the 8-channel monitor. Overall, it's more safe to report maximum profiles to the user space, so that the user can at least be able to do 8-channel playback if he want to. This patch is tested OK on G45/HDMI, IbexPeak/HDMI and IvyBridge/HDMI+DP. Minor imperfection: the GEN6_AUD_CNTL_ST/DIP_Port_Select field always reads 0 (reserved). Without knowing the port number, I worked it around by setting the ELD_valid bit for ALL the three ports. It's tested to not be a problem, because the audio driver will find invalid ELD data and hence rightfully abort, even when it sees the ELD_valid indicator. Thanks to Zhenyu and Bossart for a lot of valuable help and testing. CC: Zhao Yakui <yakui.zhao(a)intel.com> CC: Wang Zhenyu <zhenyu.z.wang(a)intel.com> CC: Jeremy Bush <contractfrombelow(a)gmail.com> CC: Christopher White <c.white(a)pulseforce.com> CC: "Bossart, Pierre-louis" <pierre-louis.bossart(a)intel.com> Signed-off-by: Ben Skeggs <bskeggs(a)redhat.com> Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- drivers/gpu/drm/drm_edid.c | 171 +++++++++++++++++++++++++ drivers/gpu/drm/i915/i915_drv.h | 2 drivers/gpu/drm/i915/i915_reg.h | 25 +++ drivers/gpu/drm/i915/intel_display.c | 132 +++++++++++++++++++ drivers/gpu/drm/i915/intel_dp.c | 6 drivers/gpu/drm/i915/intel_drv.h | 2 drivers/gpu/drm/i915/intel_hdmi.c | 3 drivers/gpu/drm/i915/intel_modes.c | 2 include/drm/drm_crtc.h | 9 + include/drm/drm_edid.h | 9 + 10 files changed, 359 insertions(+), 2 deletions(-) --- linux-next.orig/drivers/gpu/drm/drm_edid.c 2011-09-02 15:59:35.000000000 +0800 +++ linux-next/drivers/gpu/drm/drm_edid.c 2011-09-02 15:59:40.000000000 +0800 @@ -1319,6 +1319,7 @@ add_detailed_modes(struct drm_connector #define HDMI_IDENTIFIER 0x000C03 #define AUDIO_BLOCK 0x01 #define VENDOR_BLOCK 0x03 +#define SPEAKER_BLOCK 0x04 #define EDID_BASIC_AUDIO (1 << 6) /** @@ -1347,6 +1348,176 @@ u8 *drm_find_cea_extension(struct edid * } EXPORT_SYMBOL(drm_find_cea_extension); +static void +parse_hdmi_vsdb(struct drm_connector *connector, uint8_t *db) +{ + connector->eld[5] |= (db[6] >> 7) << 1; /* Supports_AI */ + + connector->dvi_dual = db[6] & 1; + connector->max_tmds_clock = db[7] * 5; + + connector->latency_present[0] = db[8] >> 7; + connector->latency_present[1] = (db[8] >> 6) & 1; + connector->video_latency[0] = db[9]; + connector->audio_latency[0] = db[10]; + connector->video_latency[1] = db[11]; + connector->audio_latency[1] = db[12]; + + DRM_LOG_KMS("HDMI: DVI dual %d, " + "max TMDS clock %d, " + "latency present %d %d, " + "video latency %d %d, " + "audio latency %d %d\n", + connector->dvi_dual, + connector->max_tmds_clock, + (int) connector->latency_present[0], + (int) connector->latency_present[1], + connector->video_latency[0], + connector->video_latency[1], + connector->audio_latency[0], + connector->audio_latency[1]); +} + +static void +monitor_name(struct detailed_timing *t, void *data) +{ + if (t->data.other_data.type == EDID_DETAIL_MONITOR_NAME) + *(u8 **)data = t->data.other_data.data.str.str; +} + +/** + * drm_edid_to_eld - build ELD from EDID + * @connector: connector corresponding to the HDMI/DP sink + * @edid: EDID to parse + * + * Fill the ELD (EDID-Like Data) buffer for passing to the audio driver. + * Some ELD fields are left to the graphics driver caller: + * - Conn_Type + * - HDCP + * - Port_ID + */ +void drm_edid_to_eld(struct drm_connector *connector, struct edid *edid) +{ + uint8_t *eld = connector->eld; + u8 *cea; + u8 *name; + u8 *db; + int sad_count = 0; + int mnl; + int dbl; + + memset(eld, 0, sizeof(connector->eld)); + + cea = drm_find_cea_extension(edid); + if (!cea) { + DRM_DEBUG_KMS("ELD: no CEA Extension found\n"); + return; + } + + name = NULL; + drm_for_each_detailed_block((u8 *)edid, monitor_name, &name); + for (mnl = 0; name && mnl < 13; mnl++) { + if (name[mnl] == 0x0a) + break; + eld[20 + mnl] = name[mnl]; + } + eld[4] = (cea[1] << 5) | mnl; + DRM_DEBUG_KMS("ELD monitor %s\n", eld + 20); + + eld[0] = 2 << 3; /* ELD version: 2 */ + + eld[16] = edid->mfg_id[0]; + eld[17] = edid->mfg_id[1]; + eld[18] = edid->prod_code[0]; + eld[19] = edid->prod_code[1]; + + for (db = cea + 4; db < cea + cea[2]; db += dbl + 1) { + dbl = db[0] & 0x1f; + + switch ((db[0] & 0xe0) >> 5) { + case AUDIO_BLOCK: /* Audio Data Block, contains SADs */ + sad_count = dbl / 3; + memcpy(eld + 20 + mnl, &db[1], dbl); + break; + case SPEAKER_BLOCK: /* Speaker Allocation Data Block */ + eld[7] = db[1]; + break; + case VENDOR_BLOCK: + /* HDMI Vendor-Specific Data Block */ + if (db[1] == 0x03 && db[2] == 0x0c && db[3] == 0) + parse_hdmi_vsdb(connector, db); + break; + default: + break; + } + } + eld[5] |= sad_count << 4; + eld[2] = (20 + mnl + sad_count * 3 + 3) / 4; + + DRM_DEBUG_KMS("ELD size %d, SAD count %d\n", (int)eld[2], sad_count); +} +EXPORT_SYMBOL(drm_edid_to_eld); + +/** + * drm_av_sync_delay - HDMI/DP sink audio-video sync delay in milli-seconds + * @connector: connector associated with the HDMI/DP sink + * @mode: the display mode + */ +int drm_av_sync_delay(struct drm_connector *connector, + struct drm_display_mode *mode) +{ + int i = !!(mode->flags & DRM_MODE_FLAG_INTERLACE); + int a, v; + + if (!connector->latency_present[0]) + return 0; + if (!connector->latency_present[1]) + i = 0; + + a = connector->audio_latency[i]; + v = connector->video_latency[i]; + + /* + * HDMI/DP sink doesn't support audio or video? + */ + if (a == 255 || v == 255) + return 0; + + /* + * Convert raw edid values to milli-seconds. + * Treat unknown latency as 0ms. + */ + if (a) + a = min(2 * (a - 1), 500); + if (v) + v = min(2 * (v - 1), 500); + + return max(v - a, 0); +} +EXPORT_SYMBOL(drm_av_sync_delay); + +/** + * drm_select_eld - select one ELD from multiple HDMI/DP sinks + * @encoder: the encoder just changed display mode + * @mode: the adjusted display mode + * + * It's possible for one encoder to be associated with multiple HDMI/DP sinks. + * The policy is now hard coded to simply use the first HDMI/DP sink's ELD. + */ +struct drm_connector *drm_select_eld(struct drm_encoder *encoder, + struct drm_display_mode *mode) +{ + struct drm_connector *connector; + struct drm_device *dev = encoder->dev; + + list_for_each_entry(connector, &dev->mode_config.connector_list, head) + if (connector->encoder == encoder && connector->eld[0]) + return connector; + + return NULL; +} +EXPORT_SYMBOL(drm_select_eld); + /** * drm_detect_hdmi_monitor - detect whether monitor is hdmi. * @edid: monitor EDID information --- linux-next.orig/drivers/gpu/drm/i915/intel_hdmi.c 2011-09-02 15:59:31.000000000 +0800 +++ linux-next/drivers/gpu/drm/i915/intel_hdmi.c 2011-09-02 15:59:40.000000000 +0800 @@ -245,8 +245,11 @@ static void intel_hdmi_mode_set(struct d sdvox |= HDMI_MODE_SELECT; if (intel_hdmi->has_audio) { + DRM_DEBUG_DRIVER("Enabling HDMI audio on pipe %c\n", + pipe_name(intel_crtc->pipe)); sdvox |= SDVO_AUDIO_ENABLE; sdvox |= SDVO_NULL_PACKETS_DURING_VSYNC; + intel_write_eld(encoder, adjusted_mode); } if (intel_crtc->pipe == 1) { --- linux-next.orig/include/drm/drm_edid.h 2011-09-02 15:59:31.000000000 +0800 +++ linux-next/include/drm/drm_edid.h 2011-09-02 15:59:40.000000000 +0800 @@ -230,4 +230,13 @@ struct edid { #define EDID_PRODUCT_ID(e) ((e)->prod_code[0] | ((e)->prod_code[1] << 8)) +struct drm_encoder; +struct drm_connector; +struct drm_display_mode; +void drm_edid_to_eld(struct drm_connector *connector, struct edid *edid); +int drm_av_sync_delay(struct drm_connector *connector, + struct drm_display_mode *mode); +struct drm_connector *drm_select_eld(struct drm_encoder *encoder, + struct drm_display_mode *mode); + #endif /* __DRM_EDID_H__ */ --- linux-next.orig/drivers/gpu/drm/i915/intel_display.c 2011-09-02 15:59:31.000000000 +0800 +++ linux-next/drivers/gpu/drm/i915/intel_display.c 2011-09-02 15:59:40.000000000 +0800 @@ -31,6 +31,7 @@ #include <linux/kernel.h> #include <linux/slab.h> #include <linux/vgaarb.h> +#include <drm/drm_edid.h> #include "drmP.h" #include "intel_drv.h" #include "i915_drm.h" @@ -5667,6 +5668,132 @@ static int intel_crtc_mode_set(struct dr return ret; } +static void g4x_write_eld(struct drm_connector *connector, + struct drm_crtc *crtc) +{ + struct drm_i915_private *dev_priv = connector->dev->dev_private; + uint8_t *eld = connector->eld; + uint32_t eldv; + uint32_t len; + uint32_t i; + + i = I915_READ(G4X_AUD_VID_DID); + + if (i == INTEL_AUDIO_DEVBLC || i == INTEL_AUDIO_DEVCL) + eldv = G4X_ELDV_DEVCL_DEVBLC; + else + eldv = G4X_ELDV_DEVCTG; + + i = I915_READ(G4X_AUD_CNTL_ST); + i &= ~(eldv | G4X_ELD_ADDR); + len = (i >> 9) & 0x1f; /* ELD buffer size */ + I915_WRITE(G4X_AUD_CNTL_ST, i); + + if (!eld[0]) + return; + + len = min_t(uint8_t, eld[2], len); + DRM_DEBUG_DRIVER("ELD size %d\n", len); + for (i = 0; i < len; i++) + I915_WRITE(G4X_HDMIW_HDMIEDID, *((uint32_t *)eld + i)); + + i = I915_READ(G4X_AUD_CNTL_ST); + i |= eldv; + I915_WRITE(G4X_AUD_CNTL_ST, i); +} + +static void ironlake_write_eld(struct drm_connector *connector, + struct drm_crtc *crtc) +{ + struct drm_i915_private *dev_priv = connector->dev->dev_private; + uint8_t *eld = connector->eld; + uint32_t eldv; + uint32_t i; + int len; + int hdmiw_hdmiedid; + int aud_cntl_st; + int aud_cntrl_st2; + + if (IS_IVYBRIDGE(connector->dev)) { + hdmiw_hdmiedid = GEN7_HDMIW_HDMIEDID_A; + aud_cntl_st = GEN7_AUD_CNTRL_ST_A; + aud_cntrl_st2 = GEN7_AUD_CNTRL_ST2; + } else { + hdmiw_hdmiedid = GEN6_HDMIW_HDMIEDID_A; + aud_cntl_st = GEN6_AUD_CNTL_ST_A; + aud_cntrl_st2 = GEN6_AUD_CNTL_ST2; + } + + i = to_intel_crtc(crtc)->pipe; + hdmiw_hdmiedid += i * 0x100; + aud_cntl_st += i * 0x100; + + DRM_DEBUG_DRIVER("ELD on pipe %c\n", pipe_name(i)); + + i = I915_READ(aud_cntl_st); + i = (i >> 29) & 0x3; /* DIP_Port_Select, 0x1 = PortB */ + if (!i) { + DRM_DEBUG_DRIVER("Audio directed to unknown port\n"); + /* operate blindly on all ports */ + eldv = GEN6_ELD_VALIDB; + eldv |= GEN6_ELD_VALIDB << 4; + eldv |= GEN6_ELD_VALIDB << 8; + } else { + DRM_DEBUG_DRIVER("ELD on port %c\n", 'A' + i); + eldv = GEN6_ELD_VALIDB << ((i - 1) * 4); + } + + i = I915_READ(aud_cntrl_st2); + i &= ~eldv; + I915_WRITE(aud_cntrl_st2, i); + + if (!eld[0]) + return; + + if (intel_pipe_has_type(crtc, INTEL_OUTPUT_DISPLAYPORT)) { + DRM_DEBUG_DRIVER("ELD: DisplayPort detected\n"); + eld[5] |= (1 << 2); /* Conn_Type, 0x1 = DisplayPort */ + } + + i = I915_READ(aud_cntl_st); + i &= ~GEN6_ELD_ADDRESS; + I915_WRITE(aud_cntl_st, i); + + len = min_t(uint8_t, eld[2], 21); /* 84 bytes of hw ELD buffer */ + DRM_DEBUG_DRIVER("ELD size %d\n", len); + for (i = 0; i < len; i++) + I915_WRITE(hdmiw_hdmiedid, *((uint32_t *)eld + i)); + + i = I915_READ(aud_cntrl_st2); + i |= eldv; + I915_WRITE(aud_cntrl_st2, i); +} + +void intel_write_eld(struct drm_encoder *encoder, + struct drm_display_mode *mode) +{ + struct drm_crtc *crtc = encoder->crtc; + struct drm_connector *connector; + struct drm_device *dev = encoder->dev; + struct drm_i915_private *dev_priv = dev->dev_private; + + dump_stack(); + connector = drm_select_eld(encoder, mode); + if (!connector) + return; + + DRM_DEBUG_DRIVER("ELD on [CONNECTOR:%d:%s], [ENCODER:%d:%s]\n", + connector->base.id, + drm_get_connector_name(connector), + connector->encoder->base.id, + drm_get_encoder_name(connector->encoder)); + + connector->eld[6] = drm_av_sync_delay(connector, mode) / 2; + + if (dev_priv->display.write_eld) + dev_priv->display.write_eld(connector, crtc); +} + /** Loads the palette/gamma unit for the CRTC with the prepared values */ void intel_crtc_load_lut(struct drm_crtc *crtc) { @@ -8183,6 +8310,7 @@ static void intel_init_display(struct dr } dev_priv->display.fdi_link_train = ironlake_fdi_link_train; dev_priv->display.init_clock_gating = ironlake_init_clock_gating; + dev_priv->display.write_eld = ironlake_write_eld; } else if (IS_GEN6(dev)) { if (SNB_READ_WM0_LATENCY()) { dev_priv->display.update_wm = sandybridge_update_wm; @@ -8193,6 +8321,7 @@ static void intel_init_display(struct dr } dev_priv->display.fdi_link_train = gen6_fdi_link_train; dev_priv->display.init_clock_gating = gen6_init_clock_gating; + dev_priv->display.write_eld = ironlake_write_eld; } else if (IS_IVYBRIDGE(dev)) { /* FIXME: detect B0+ stepping and use auto training */ dev_priv->display.fdi_link_train = ivb_manual_fdi_link_train; @@ -8204,7 +8333,7 @@ static void intel_init_display(struct dr dev_priv->display.update_wm = NULL; } dev_priv->display.init_clock_gating = ivybridge_init_clock_gating; - + dev_priv->display.write_eld = ironlake_write_eld; } else dev_priv->display.update_wm = NULL; } else if (IS_PINEVIEW(dev)) { @@ -8224,6 +8353,7 @@ static void intel_init_display(struct dr dev_priv->display.update_wm = pineview_update_wm; dev_priv->display.init_clock_gating = gen3_init_clock_gating; } else if (IS_G4X(dev)) { + dev_priv->display.write_eld = g4x_write_eld; dev_priv->display.update_wm = g4x_update_wm; dev_priv->display.init_clock_gating = g4x_init_clock_gating; } else if (IS_GEN4(dev)) { --- linux-next.orig/drivers/gpu/drm/i915/i915_reg.h 2011-09-02 15:59:31.000000000 +0800 +++ linux-next/drivers/gpu/drm/i915/i915_reg.h 2011-09-02 15:59:40.000000000 +0800 @@ -3470,4 +3470,29 @@ #define GEN6_PCODE_DATA 0x138128 #define GEN6_PCODE_FREQ_IA_RATIO_SHIFT 8 +#define G4X_AUD_VID_DID 0x62020 +#define INTEL_AUDIO_DEVCL 0x808629FB +#define INTEL_AUDIO_DEVBLC 0x80862801 +#define INTEL_AUDIO_DEVCTG 0x80862802 + +#define G4X_AUD_CNTL_ST 0x620B4 +#define G4X_ELDV_DEVCL_DEVBLC (1 << 13) +#define G4X_ELDV_DEVCTG (1 << 14) +#define G4X_ELD_ADDR (0xf << 5) +#define G4X_ELD_ACK (1 << 4) +#define G4X_HDMIW_HDMIEDID 0x6210C + +#define GEN6_HDMIW_HDMIEDID_A 0xE2050 +#define GEN6_AUD_CNTL_ST_A 0xE20B4 +#define GEN6_ELD_BUFFER_SIZE (0x1f << 10) +#define GEN6_ELD_ADDRESS (0x1f << 5) +#define GEN6_ELD_ACK (1 << 4) +#define GEN6_AUD_CNTL_ST2 0xE20C0 +#define GEN6_ELD_VALIDB (1 << 0) +#define GEN6_CP_READYB (1 << 1) + +#define GEN7_HDMIW_HDMIEDID_A 0xE5050 +#define GEN7_AUD_CNTRL_ST_A 0xE50B4 +#define GEN7_AUD_CNTRL_ST2 0xE50C0 + #endif /* _I915_REG_H_ */ --- linux-next.orig/drivers/gpu/drm/i915/intel_drv.h 2011-09-02 15:59:31.000000000 +0800 +++ linux-next/drivers/gpu/drm/i915/intel_drv.h 2011-09-02 15:59:40.000000000 +0800 @@ -380,4 +380,6 @@ extern void intel_fb_output_poll_changed extern void intel_fb_restore_mode(struct drm_device *dev); extern void intel_init_clock_gating(struct drm_device *dev); +extern void intel_write_eld(struct drm_encoder *encoder, + struct drm_display_mode *mode); #endif /* __INTEL_DRV_H__ */ --- linux-next.orig/drivers/gpu/drm/i915/intel_modes.c 2011-09-02 15:59:31.000000000 +0800 +++ linux-next/drivers/gpu/drm/i915/intel_modes.c 2011-09-02 15:59:40.000000000 +0800 @@ -26,6 +26,7 @@ #include <linux/slab.h> #include <linux/i2c.h> #include <linux/fb.h> +#include <drm/drm_edid.h> #include "drmP.h" #include "intel_drv.h" #include "i915_drv.h" @@ -74,6 +75,7 @@ int intel_ddc_get_modes(struct drm_conne if (edid) { drm_mode_connector_update_edid_property(connector, edid); ret = drm_add_edid_modes(connector, edid); + drm_edid_to_eld(connector, edid); connector->display_info.raw_edid = NULL; kfree(edid); } --- linux-next.orig/include/drm/drm_crtc.h 2011-09-02 15:59:31.000000000 +0800 +++ linux-next/include/drm/drm_crtc.h 2011-09-02 15:59:40.000000000 +0800 @@ -466,6 +466,8 @@ enum drm_connector_force { /* DACs should rarely do this without a lot of testing */ #define DRM_CONNECTOR_POLL_DISCONNECT (1 << 2) +#define MAX_ELD_BYTES 128 + /** * drm_connector - central DRM connector control structure * @crtc: CRTC this connector is currently connected to, NULL if none @@ -523,6 +525,13 @@ struct drm_connector { uint32_t force_encoder_id; struct drm_encoder *encoder; /* currently active encoder */ + /* EDID bits */ + uint8_t eld[MAX_ELD_BYTES]; + bool dvi_dual; + int max_tmds_clock; /* in MHz */ + bool latency_present[2]; + int video_latency[2]; /* [0]: progressive, [1]: interlaced */ + int audio_latency[2]; int null_edid_counter; /* needed to workaround some HW bugs where we get all 0s */ }; --- linux-next.orig/drivers/gpu/drm/i915/intel_dp.c 2011-09-02 15:59:31.000000000 +0800 +++ linux-next/drivers/gpu/drm/i915/intel_dp.c 2011-09-02 15:59:40.000000000 +0800 @@ -773,8 +773,12 @@ intel_dp_mode_set(struct drm_encoder *en intel_dp->DP |= DP_PORT_WIDTH_4; break; } - if (intel_dp->has_audio) + if (intel_dp->has_audio) { + DRM_DEBUG_DRIVER("Enabling DP audio on pipe %c\n", + pipe_name(intel_crtc->pipe)); intel_dp->DP |= DP_AUDIO_OUTPUT_ENABLE; + intel_write_eld(encoder, adjusted_mode); + } memset(intel_dp->link_configuration, 0, DP_LINK_CONFIGURATION_SIZE); intel_dp->link_configuration[0] = intel_dp->link_bw; --- linux-next.orig/drivers/gpu/drm/i915/i915_drv.h 2011-09-02 15:59:31.000000000 +0800 +++ linux-next/drivers/gpu/drm/i915/i915_drv.h 2011-09-02 15:59:40.000000000 +0800 @@ -209,6 +209,8 @@ struct drm_i915_display_funcs { struct drm_display_mode *adjusted_mode, int x, int y, struct drm_framebuffer *old_fb); + void (*write_eld)(struct drm_connector *connector, + struct drm_crtc *crtc); void (*fdi_link_train)(struct drm_crtc *crtc); void (*init_clock_gating)(struct drm_device *dev); void (*init_pch_clock_gating)(struct drm_device *dev);

3 6

Re: [alsa-devel] [PATCH v5] drm/i915: pass ELD to HDMI/DP audio driver
by Wu Fengguang 05 Sep '11

05 Sep '11

On Sun, Sep 04, 2011 at 08:08:37PM +0800, Chris Wilson wrote: > On Sun, 4 Sep 2011 05:15:10 +0800, Wu Fengguang <fengguang.wu(a)intel.com> wrote: > > Changes from v4: remove a debug call to dump_stack(). > > Thanks to Bossart for catching this! > > --- > > > > Add ELD support for Intel Eaglelake, IbexPeak/Ironlake, > > SandyBridge/CougarPoint and IvyBridge/PantherPoint chips. > > > > ELD (EDID-Like Data) describes to the HDMI/DP audio driver the audio > > capabilities of the plugged monitor. It's built and passed to audio > > driver in 2 steps: > > > > (1) at get_modes time, parse EDID and save ELD to drm_connector.eld[] > > > > (2) at mode_set time, write drm_connector.eld[] to the Transcoder's hw > > ELD buffer and set the ELD_valid bit to inform HDMI/DP audio driver > > > > ELD selection policy: it's possible for one encoder to be associated > > with multiple connectors (ie. monitors), in which case the first found > > ELD will be used. This policy may not be suitable for all users, but > > let's start it simple first. > > > > The impact of ELD selection policy: assume there are two monitors, one > > supports stereo playback and the other has 8-channel output; cloned > > display mode is used, so that the two monitors are associated with the > > same internal encoder. If only the stereo playback capability is reported, > > the user won't be able to start 8-channel playback; if the 8-channel ELD > > is reported, then user space applications may send 8-channel samples > > down, however the user may actually be listening to the 2-channel > > monitor and not connecting speakers to the 8-channel monitor. Overall, > > it's more safe to report maximum profiles to the user space, so that > > the user can at least be able to do 8-channel playback if he want to. > > > > This patch is tested OK on G45/HDMI, IbexPeak/HDMI and IvyBridge/HDMI+DP. > > > > Minor imperfection: the GEN6_AUD_CNTL_ST/DIP_Port_Select field always > > reads 0 (reserved). Without knowing the port number, I worked it around > > by setting the ELD_valid bit for ALL the three ports. It's tested to not > > be a problem, because the audio driver will find invalid ELD data and > > hence rightfully abort, even when it sees the ELD_valid indicator. > > > > > --- linux-next.orig/drivers/gpu/drm/i915/intel_display.c 2011-09-02 15:59:31.000000000 +0800 > > +++ linux-next/drivers/gpu/drm/i915/intel_display.c 2011-09-04 05:11:44.000000000 +0800 > > +static void ironlake_write_eld(struct drm_connector *connector, > > + struct drm_crtc *crtc) > > +{ > > + struct drm_i915_private *dev_priv = connector->dev->dev_private; > > + uint8_t *eld = connector->eld; > > + uint32_t eldv; > > + uint32_t i; > > + int len; > > + int hdmiw_hdmiedid; > > + int aud_cntl_st; > > + int aud_cntrl_st2; > > + > > + if (IS_IVYBRIDGE(connector->dev)) { > > + hdmiw_hdmiedid = GEN7_HDMIW_HDMIEDID_A; > > + aud_cntl_st = GEN7_AUD_CNTRL_ST_A; > > + aud_cntrl_st2 = GEN7_AUD_CNTRL_ST2; > > + } else { > > + hdmiw_hdmiedid = GEN6_HDMIW_HDMIEDID_A; > > + aud_cntl_st = GEN6_AUD_CNTL_ST_A; > > + aud_cntrl_st2 = GEN6_AUD_CNTL_ST2; > > + } > > This register naming is inconsistent with its intent. If these registers > were indeed introduced on Ironlake as you seem to imply by using them > with Ironlake, you should label them as GEN5 and not GEN6. Yeah, I admit that I'm a bit confused in choosing the exact names. I'll fix that. > This patch should be split between adding the core drm functionality to > build the ELD and the introduction of i915 support. OK. I didn't do this because I was not sure if it's OK to just add the drm_*() functions without any code to call them.. Thanks, Fengguang

1 0

Re: [alsa-devel] [PATCH] ASoC: rt5631: first commit
by Mark Brown 04 Sep '11

04 Sep '11

On Thu, Sep 01, 2011 at 08:21:14PM +0800, johnnyhsu(a)realtek.com wrote: > Subject: [PATCH] ASoC: rt5631: first commit "Add driver for..." or something. > +static int rt5631_volatile_register(struct snd_soc_codec *codec, > + unsigned int reg) > +{ > + switch (reg) { > + case RT5631_VENDOR_ID: > + case RT5631_VENDOR_ID1: > + case RT5631_VENDOR_ID2: > + return 0; > + default: > + return 1; > + } > +} This looks to be exactly the wrong way round - the only registers we can cache are the ID registers. > +/* MIC Boost Volume */ > +static const char *rt5631_mic_boost[] = {"Bypass", "+20db", "+24db", "+30db", > + "+35db", "+40db", "+44db", "+50db", "+52db"}; TLV for volume information. > + rt5631->bclk_rate = snd_soc_params_to_bclk(params); > + if (rt5631->bclk_rate < 0) { > + dev_err(codec->dev, "Unsupported BCLK rate: %d\n", > + rt5631->bclk_rate); > + return rt5631->bclk_rate; > + } The error message here is inacurate, there was an error obtaining the BCLK rate rather than a problem supporting it. > +static int rt5631_codec_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, > + int source, unsigned int freq_in, unsigned int freq_out) > +{ > + struct snd_soc_codec *codec = codec_dai->codec; > + struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec); > + int i, ret = -EINVAL; > + > + dev_dbg(codec->dev, "enter %s\n", __func__); > + > + if (!freq_in || !freq_out) { > + dev_dbg(codec->dev, "PLL disabled\n"); > + return -EINVAL; > + } This shouldn't be an error condition, disabling the PLL is a perfectly normal thing to do. > + if (rt5631->master) { > + for (i = 0; i < ARRAY_SIZE(codec_master_pll_div); i++) > + if (freq_in == codec_master_pll_div[i].pll_in && Since the PLL configuration depends on the master/slave configuration you should either warn or reconfigure if the PLL is set up and the user changes master/slave configuration. > +/** > + * rt5631_index_reg_show - sysfs file for dumping index registers of 2nd layer > + */ > +static ssize_t rt5631_index_reg_show(struct device *dev, > + struct device_attribute *attr, char *buf) > +{ No, use the standard facilities. Please don't ignore review comments, it's not helpful.

1 0

Re: [alsa-devel] [PATCH] ASoC: sn95031: Fix the logic to find free channel
by Mark Brown 04 Sep '11

04 Sep '11

On Sat, Sep 03, 2011 at 01:41:47PM +0800, Axel Lin wrote: > In the case of no free channel available, > current implementation returns 0 instead of negative errno. > > This patch fixes the logic to return -EINVAL if no free channel available. Applied, thanks.

1 0

Re: [alsa-devel] [PATCH] ASoC: codecs: Remove unused mutex
by Mark Brown 04 Sep '11

04 Sep '11

On Sun, Sep 04, 2011 at 10:48:09PM +0800, Axel Lin wrote: > 2011/9/4 Mark Brown <broonie(a)opensource.wolfsonmicro.com>: > > This has already been done for wm9090, can you resubmit with just the > > alc5623 change please? > Ok. A new patch "ASoC: alc5623: Remove unused mutex" is on the way. Thanks. The patch has been sitting there for a little while but it's not on kernel.org due to the lockdown after the compromise - until we can start pushing updates again there's a risk of overlaps like this.

1 0

[alsa-devel] [3.1-rc4] sound regression on Realtek ALC260
by Alessandro Guido 04 Sep '11

04 Sep '11

With 3.1-rc4 Skype makes very unpleasant noises when playing event sounds or voice. Sometimes a reboot may be able to fix it, but with with 3.0 it always worked fine. Some new warnings are shown in dmesg for 3.1-rc4 related to alsa: hda-intel: Invalid position buffer, using LPIB read method instead. hda-intel: Invalid position buffer, using LPIB read method instead. hda-intel: IRQ timing workaround is activated for card #0. Suggest a bigger bdl_pos_adj. I'm attaching the output of alsa-info utility.

1 0

Re: [alsa-devel] [PATCH] ASoC: codecs: Remove unused mutex
by Mark Brown 04 Sep '11

04 Sep '11

On Sun, Sep 04, 2011 at 08:52:31PM +0800, Axel Lin wrote: > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> > Cc: Arnaud Patard <arnaud.patard(a)rtp-net.org> > Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> > --- > sound/soc/codecs/alc5623.c | 2 -- > sound/soc/codecs/wm9090.c | 2 -- This has already been done for wm9090, can you resubmit with just the alc5623 change please?

1 0

Re: [alsa-devel] [PATCH] ASoC: sn95031: Fix the logic to find free channel
by Girdwood, Liam 03 Sep '11

03 Sep '11

On 3 September 2011 06:41, Axel Lin <axel.lin(a)gmail.com> wrote: > In the case of no free channel available, > current implementation returns 0 instead of negative errno. > > This patch fixes the logic to return -EINVAL if no free channel available. > > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> > --- > sound/soc/codecs/sn95031.c | 8 +++----- > 1 files changed, 3 insertions(+), 5 deletions(-) > > diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c > index 84ffdeb..b4f1cb4 100644 > --- a/sound/soc/codecs/sn95031.c > +++ b/sound/soc/codecs/sn95031.c > @@ -79,7 +79,7 @@ static void configure_adc(struct snd_soc_codec > *sn95031_codec, int val) > */ > static int find_free_channel(struct snd_soc_codec *sn95031_codec) > { > - int ret = 0, i, value; > + int i, value; > > /* check whether ADC is enabled */ > value = snd_soc_read(sn95031_codec, SN95031_ADC1CNTL1); > @@ -91,12 +91,10 @@ static int find_free_channel(struct snd_soc_codec > *sn95031_codec) > for (i = 0; i < SN95031_ADC_CHANLS_MAX; i++) { > value = snd_soc_read(sn95031_codec, > SN95031_ADC_CHNL_START_ADDR + i); > - if (value & SN95031_STOPBIT_MASK) { > - ret = i; > + if (value & SN95031_STOPBIT_MASK) > break; > - } > } > - return (ret > SN95031_ADC_LOOP_MAX) ? (-EINVAL) : ret; > + return (i == SN95031_ADC_CHANLS_MAX) ? (-EINVAL) : i; > } > > /* Initialize the ADC for reading micbias values. Can sleep. */ > -- > 1.7.4.1 > > > > Acked-by: Liam Girdwood <lrg(a)ti.com> Btw, this should be posted to alsa-devel.

2 1

[alsa-devel] Lost dB volumes on USB headset between 2.6.38 and 3.0
by David Henningsson 03 Sep '11

03 Sep '11

Hmm, I and Colin Guthrie have found a regression between kernels 2.6.38 and 3.0. When running "amixer -D hw:handset" for this device under 2.6.38, the attached file is returned. Under 3.0, the dB information is missing. This can't be intentional, can it? -- David Henningsson http://launchpad.net/~diwic

5 13

[alsa-devel] A debug version of the ALSA lib?
by Olivier Guillion - Myriad 02 Sep '11

02 Sep '11

Hi, I have a problem with the ALSA library. Calling snd_device_name_hint() crashes sometimes, and I can't understand why. It might be due to a previous memory corruption in my program, a problem with my sound card driver, or a bug in the library, I don't know. I tried to debug it in assembler, and found that the library is browsing a list of devices, gets the pointer to each device name, and in the 2nd device, I get the four letters "Play" instead of a valid pointer. To be able to go further, I'd need a debug version of the ALSA library, including variable names and source lines, which could let me find more easily what goes wrong and when. Is it possible to build such a library from the alsa-lib-1.0.24.1 sources? Or should I download and install another specific package? Thank you, Olivier

6 15

Re: [alsa-devel] Can ALSA invoke a Callback ?
by Clemens Ladisch 02 Sep '11

02 Sep '11

Ru Vuott wrote: > hData = OPEN "/dev/snd/midiC2D0" for READ Watch > > Do you think this solution could interfere and give problems to ALSA ? Yes; this opens the device exclusively. The ALSA sequencer would be able to allow multiple applications. > The rest of my program (for sending midi event, etcetera) continue to use ALSA. How are you using ALSA from Gambas? Is there some ALSA module that could be extended to ALSA watches? Regards, Clemens

1 0

[alsa-devel] [RFC PATCH 0/3] If an IRQ is a GPIO, request and configure it
by Stephen Warren 02 Sep '11

02 Sep '11

In http://www.spinics.net/lists/linux-tegra/msg01731.html, Mark Brown pointed out that it was a little silly forcing every board or driver to gpio_request() a GPIO that is later converted to an IRQ, and passed to request_irq. The first patch in this series instead makes the core IRQ code perform these calls when appropriate, to avoid duplicating it everywhere. However, this change has the potential for significant regressions; at least some drivers are already calling gpio_request for GPIOs that are also used as IRQs. This then causes the gpio_request inside the core IRQ code to fail, which causes functional regressions. I'm not sure how wide- spread this issue is, but in testing on NVIDIA Tegra, I found two instances that needed to be fixed. Perhaps a failure of gpio_request in the core IRQ code should trigger a WARN rather than returning an error, to give a grace period for conversion of other code? Stephen Warren (3): irq: If an IRQ is a GPIO, request and configure it mmc: tegra: Don't gpio_request GPIOs used as IRQs. ASoC: jack_add_gpios: Don't gpio_request GPIOs used as IRQs. drivers/mmc/host/sdhci-tegra.c | 8 -------- kernel/irq/manage.c | 25 +++++++++++++++++++++++-- sound/soc/soc-jack.c | 13 +------------ 3 files changed, 24 insertions(+), 22 deletions(-)

5 20

Re: [alsa-devel] [RFC 0/5] compress: add compress data API to ALSA kernel
by Mark Brown 02 Sep '11

02 Sep '11

On Fri, Sep 02, 2011 at 09:04:42AM -0500, Pierre-Louis Bossart wrote: > Not yet. I wanted to have more visibility into alsa-lib use for > Android before doing into changes there. > Let's talk about it at LPC. FWIW and the benefit of the list alsa-lib won't fly for adoption by Google as part of the standard Android system unless it gets relicensed, though obviously other people are happily using it in production and there's systems other than Android out there. tinyalsa is probably the way forward for Google.

1 0

Re: [alsa-devel] [PATCH] ASoC: Add missing platform_device_put in raumfeld_audio_init error path
by Girdwood, Liam 02 Sep '11

02 Sep '11

On 2 September 2011 14:47, Axel Lin <axel.lin(a)gmail.com> wrote: > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> > --- > sound/soc/pxa/raumfeld.c | 4 +++- > 1 files changed, 3 insertions(+), 1 deletions(-) > > diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c > index 1a591f1..b899a3b 100644 > --- a/sound/soc/pxa/raumfeld.c > +++ b/sound/soc/pxa/raumfeld.c > @@ -306,8 +306,10 @@ static int __init raumfeld_audio_init(void) > &snd_soc_raumfeld_connector); > > ret = platform_device_add(raumfeld_audio_device); > - if (ret < 0) > + if (ret < 0) { > + platform_device_put(raumfeld_audio_device); > return ret; > + } > > raumfeld_enable_audio(true); > return 0; > -- > 1.7.4.1 > Acked-by: Liam Girdwood <lrg(a)ti.com>

1 0

[alsa-devel] submitting patch: specify destination kernel version(s)
by Torsten Schenk 02 Sep '11

02 Sep '11

Hi, I'm currently preparing a patch for the kernel, I want it to be included from 3.0 on, but not in versions before. I once got a hint to put Cc: stable(a)kernel.org but that seems not to do the trick. How could it be achieved, that the patch only appears in 3.0 (and later)? Greets, Torsten

4 3

Re: [alsa-devel] [OctDev] Updated ALSA and a new JACK package for playing and recording audio
by Fredrik Lingvall 02 Sep '11

02 Sep '11

Miguel, I have updated the alsa package now and I don't have any build issues on my machine: octave:3> pkg install aaudio-0.2.1.tar.gz warning: creating installation directory /usr/share/octave/packages octave:4> Check that the clock is set correctly on your machine (is ntp-client running?) Also, test to do a octave:4> pkg uninstall aaudio and then install aaudio again. You should have: octave:4> pkg list Package Name | Version | Installation directory --------------+---------+----------------------- aaudio *| 0.2.1 | /usr/share/octave/packages/aaudio-0.2.1 after installation. /Fredrik On 09/01/11 21:53, Miguel Bazdresch wrote: > On Thu, Sep 1, 2011 at 12:29 PM, Fredrik Lingvall <fl(a)ifi.uio.no> wrote: >> Ah, I forgot to remove that line from the Makefile. >> >> You can fix it by uncompressing the tar file, edit the Makefile in the src >> folder, remove all actrecord stuff, and recomress it again. I will fix this >> tomorrow. >> >> /Fredrik > Fredrik, > > After removing all references to actrecord, I got: > > octave:1> pkg install /home/miguel/Sources/aaudio-0.2.0.tar.gz > warning: load_path: > /home/miguel/octave/aaudio-0.2.0/i686-pc-linux-gnu-api-v45+: No such > file or directory > /home/miguel/octave/aaudio-0.2.0/i686-pc-linux-gnu-api-v45+/aplayrec.oct: > failed to load: > /home/miguel/octave/aaudio-0.2.0/i686-pc-linux-gnu-api-v45+/aplayrec.oct: > undefined symbol: snd_pcm_sw_params_set_start_threshold > error: called from `gen_doc_cache>create_cache' in file > /usr/share/octave/3.4.2/m/help/gen_doc_cache.m near line 93, column 17 > error: called from: > error: /usr/share/octave/3.4.2/m/pkg/pkg.m at line 832, column 5 > error: /usr/share/octave/3.4.2/m/pkg/pkg.m at line 361, column 9 > octave:1> > > If I try to run a command I get: > > octave:1> aplay > warning: load_path: > /home/miguel/octave/aaudio-0.2.0/i686-pc-linux-gnu-api-v45+: No such > file or directory > warning: timestamp on file > /home/miguel/octave/aaudio-0.2.0/i686-pc-linux-gnu-api-v45+/aplay.oct > is in the future > error: /home/miguel/octave/aaudio-0.2.0/i686-pc-linux-gnu-api-v45+/aplay.oct: > failed to load: > /home/miguel/octave/aaudio-0.2.0/i686-pc-linux-gnu-api-v45+/aplay.oct: > cannot open shared object file: No such file or directory > octave:1> > > Cheers, >

1 0

[alsa-devel] ASoC DSP and related status
by Stephen Warren 01 Sep '11

01 Sep '11

Liam, Mark, I was recently talking to our internal audio team, extolling the virtues of writing upstreamable drivers for Tegra's audio HW. One of the big unknowns here is how to represent the Tegra DAS and AHUB modules[1] in a standard fashion, and allowing configuration via kcontrols that influence DAPM routing, rather than open-coding and/or hard-coding such policy in the ASoC machine driver. So, my questions are: * What's the status of the ASoC DSP work. I see that some of the base infra-structure has been merged into ASoC's for-next branch, but I think that's just a small portion of the work. Do you have any kind of estimate for when the whole thing will be merged? I don't see recent updates to e.g. Liams' topic/dsp or topic/dsp-upstream branches. * Back in March in another DSP-related thread, Mark mentioned that the DSP rework was mainly about configuring stuff within a device, but that Mark was working on some code to support autonomous inter-device links. I assume that Tegra's DAS/AHUB would rely on the DSP work, not the stuff Mark mentioned? See the last few paragraphs of: http://mailman.alsa-project.org/pipermail/alsa-devel/2011-March/037776.html Related, Mark also mentioned something about representing the DAS/AHUB as codecs. I'm not sure if that was meant as a stop-gap solution before the DSP work was in place, or if that's part of supporting DAAS/AHUB within the DSP infra-structure. Thanks for any kind of information! Any information here will simply help us plan for when we might be able to switch from open-coding some of the more advanced Tegra audio support to more standardized solutions. [1] Here's a very quick overview of the relevant Tegra audio HW: DAS: Part of Tegra20. Tegra20 has Digital Audio controllers (DACs) i.e. I2S controllers. It also has Digital Audio Ports (DAPs). The Digital Audio Switch (DAS ) sits between. Each DAP selects its audio output from a particular DAC's or DAP's output. Each DAC selects its audio input from a particular DAP. DAP<-> DAP is supported, with one being the master the other the slave. Note that I2S configuration (channels, sample size, I2S vs. DSP etc) is configured in the DAC not the DAP. AHUB: Part of Tegra30. Tegra30 has an interconnect called the Audio HUB (AHUB). Various devices attach to this: FIFOs to send/receive audio to CPU memory using DMA, DAMs that receive n(2?) channels from the AHUB and mix/SRC them sending the result back to the AHUB, and finally various IO controllers such as I2S and SPDIF. The AHUB is I believe a full cross-bar. In this case, the I2S formatting is configured solely within the I2S controllers, not on the other side of the AHUB as is the case with the Tegra20 DAS. FIFOs also independently determine the number of channels/bit they send/ receive. There is some limited support for channel count and bitsize conversion at each attachment point to the AHUB. I2S<->I2S loopback may be supported in HW, at least in some cases. -- nvpublic

3 5

[alsa-devel] Have multiple opens removed on one device?
by Scott Jiang 01 Sep '11

01 Sep '11

Hi Mark and Takashi, In kernel 2.6.34 I can use aplay to play audio at the same time, alsa will mix the audio. Now the second open will return device busy. So it seems this feature has been removed. What's the proper approach for this operation at present? Thanks. Scott

3 4

[alsa-devel] [PATCH utils] Fix use of $(SED).
by Michał Górny 31 Aug '11

31 Aug '11

The alsactl Makefile used 'sed $(SED)' which is going to failed when SED is actually set. Replaced that with '$(SED)', and a few 'sed' calls in alsaconf Makefile as well. Added AC_PROG_SED to configure to have it set. Fixes: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5363 Fixes: http://bugs.gentoo.org/show_bug.cgi?id=366587 Signed-off-by: Michał Górny <mgorny(a)gentoo.org> --- alsaconf/Makefile.am | 16 ++++++++-------- alsactl/Makefile.am | 2 +- configure.in | 1 + 3 files changed, 10 insertions(+), 9 deletions(-) diff --git a/alsaconf/Makefile.am b/alsaconf/Makefile.am index a11ecc5..a4f5e4a 100644 --- a/alsaconf/Makefile.am +++ b/alsaconf/Makefile.am @@ -5,13 +5,13 @@ SUBDIRS = po install-man8: @for i in $(man_MANS); do \ - ext=`echo $$i | sed -e 's/^.*\\.//'`; \ - inst=`echo $$i | sed -e 's/\\.[^.]*$$//'`; \ + ext=`echo $$i | $(SED) -e 's/^.*\\.//'`; \ + inst=`echo $$i | $(SED) -e 's/\\.[^.]*$$//'`; \ case $$inst in \ *.[a-za-z]*)\ - loc=`echo $$inst | sed -e 's/^.*\\.//'`; \ + loc=`echo $$inst | $(SED) -e 's/^.*\\.//'`; \ loc="/$$loc"; \ - inst=`echo $$inst | sed -e 's/\\.[^.]*$$//'`;; \ + inst=`echo $$inst | $(SED) -e 's/\\.[^.]*$$//'`;; \ *)\ loc="";; \ esac; \ @@ -22,13 +22,13 @@ install-man8: uninstall-man8: @for i in $(man_MANS); do \ - ext=`echo $$i | sed -e 's/^.*\\.//'`; \ - inst=`echo $$i | sed -e 's/\\.[^.]*$$//'`; \ + ext=`echo $$i | $(SED) -e 's/^.*\\.//'`; \ + inst=`echo $$i | $(SED) -e 's/\\.[^.]*$$//'`; \ case $$inst in \ *.[a-za-z]*)\ - loc=`echo $$inst | sed -e 's/^.*\\.//'`; \ + loc=`echo $$inst | $(SED) -e 's/^.*\\.//'`; \ loc="/$$loc"; \ - inst=`echo $$inst | sed -e 's/\\.[^.]*$$//'`;; \ + inst=`echo $$inst | $(SED) -e 's/\\.[^.]*$$//'`;; \ *)\ loc="";; \ esac; \ diff --git a/alsactl/Makefile.am b/alsactl/Makefile.am index 5cfc415..c434e44 100644 --- a/alsactl/Makefile.am +++ b/alsactl/Makefile.am @@ -34,7 +34,7 @@ install-data-hook: endif edit = \ - sed $(SED) -r 's,@sbindir\@,$(sbindir),g' < $< > $@ || rm $@ + $(SED) -r 's,@sbindir\@,$(sbindir),g' < $< > $@ || rm $@ alsa-store.service: alsa-store.service.in $(edit) diff --git a/configure.in b/configure.in index 9ec1fac..b28af80 100644 --- a/configure.in +++ b/configure.in @@ -28,6 +28,7 @@ dnl AC_PROG_CXX AC_PROG_INSTALL AC_PROG_MKDIR_P AC_PROG_LN_S +AC_PROG_SED AM_PATH_ALSA(1.0.16) dnl Check components -- 1.7.6.1

2 1

[alsa-devel] loading firmware to the codec not successful
by Appalayagari Sreedhar 31 Aug '11

31 Aug '11

Hi, I am working with DM365 custom board the linux kernel version is 2.6.37. I am working with a cx2070x codec which requires firmware to be loaded. I have placed the firmware file in the path. "usr/lib/firmware" it is not working, i am getting the error. cx2070x_download_firmware(): Firmware cx2070x.fw not available -2 cx2070x_init: failed to download firmware error -2 indicates file not available. I think the path is not correct, can you please suggest which path i should place the firmware file and also any configuration i need to enable in the kernel. Thank you.

2 1

[alsa-devel] Alsa timing question
by Raymond Toy 31 Aug '11

31 Aug '11

I'm working on an application where I'd like to have relatively fixed timing between calls to snd_pcm_writei. As an example, of this I used the sample pcm.c code from < http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html#a…>. I placed a simple function to write out the time (from clock_getttime) before each call to snd_pcm_writei in write_loop. With a sample rate of 48kHz and a buffer time of 128000 and period time of 42667, what I see is that most calls to snd_pcm_writei are spaced about 50 ms apart. This makes some sense, but I was expecting the calls to be about 42.67 ms apart. However, about every 6th call, the time is just 200 us or so. Why is that? Is there any simple way to make the calls spaced more closely to 42.67 ms apart? Do I need to do something more complicated to achieve this? Ray

4 12

[alsa-devel] [RFC 1/2] ASoC: imx-ssi: use dma_writecombine consistently
by Wolfram Sang 31 Aug '11

31 Aug '11

If the channel is allocated as writecombine, then mmaping it should also use writecombine. Also, add a proper device for the call. Ported from a similar fix for mach-mxs. Signed-off-by: Wolfram Sang <w.sang(a)pengutronix.de> Cc: Sascha Hauer <s.hauer(a)pengutronix.de> Cc: Liam Girdwood <lrg(a)ti.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/imx/imx-ssi.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index 10a8e27..3b8d5cd 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -357,8 +357,8 @@ int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; int ret; - ret = dma_mmap_coherent(NULL, vma, runtime->dma_area, - runtime->dma_addr, runtime->dma_bytes); + ret = dma_mmap_writecombine(substream->pcm->card->dev, vma, + runtime->dma_area, runtime->dma_addr, runtime->dma_bytes); pr_debug("%s: ret: %d %p 0x%08x 0x%08x\n", __func__, ret, runtime->dma_area, -- 1.7.5.4

4 5

[alsa-devel] [PATCH] ASoC: Allow register defaults to be larger than unsigned short
by Mark Brown 31 Aug '11

31 Aug '11

Devices that need this exist; obviously the newer regmap defaults mechanism will deal with this more happily. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 6da55a1..0fc8f15 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -622,7 +622,7 @@ struct snd_soc_codec_driver { int (*volatile_register)(struct snd_soc_codec *, unsigned int); int (*readable_register)(struct snd_soc_codec *, unsigned int); int (*writable_register)(struct snd_soc_codec *, unsigned int); - short reg_cache_size; + unsigned int reg_cache_size; short reg_cache_step; short reg_word_size; const void *reg_cache_default; -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: Allow source specification for CODEC level sysclk
by Mark Brown 31 Aug '11

31 Aug '11

Similarly to PLLs/FLLs some modern CODECs provide selectable system clock sources. When the clock is the clock for a DAI we do not usually need to identify which clock is being configured so can use clk_id for the source clock but with CODEC wide system clocks we will need to specify both the clock being configured and the source. Add a source argument to the CODEC driver set_sysclk() operation to reflect this. As this operation is not as widely used as the DAI set_sysclk() operation the change is not very invasive. We probably ought to go and make the same alternation for DAIs at some point. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 4 ++-- sound/soc/codecs/wm9081.c | 4 ++-- sound/soc/samsung/speyside.c | 2 +- sound/soc/soc-core.c | 8 +++++--- 4 files changed, 10 insertions(+), 8 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 0fc8f15..24e17be 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -276,7 +276,7 @@ enum snd_soc_pcm_subclass { }; int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, - unsigned int freq, int dir); + int source, unsigned int freq, int dir); int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source, unsigned int freq_in, unsigned int freq_out); @@ -610,7 +610,7 @@ struct snd_soc_codec_driver { /* codec wide operations */ int (*set_sysclk)(struct snd_soc_codec *codec, - int clk_id, unsigned int freq, int dir); + int clk_id, int source, unsigned int freq, int dir); int (*set_pll)(struct snd_soc_codec *codec, int pll_id, int source, unsigned int freq_in, unsigned int freq_out); diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index a469132..f32ab1e 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1120,8 +1120,8 @@ static int wm9081_digital_mute(struct snd_soc_dai *codec_dai, int mute) return 0; } -static int wm9081_set_sysclk(struct snd_soc_codec *codec, - int clk_id, unsigned int freq, int dir) +static int wm9081_set_sysclk(struct snd_soc_codec *codec, int clk_id, + int source, unsigned int freq, int dir) { struct wm9081_priv *wm9081 = snd_soc_codec_get_drvdata(codec); diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index 12e60e3..401ca22 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -223,7 +223,7 @@ static int speyside_wm9081_init(struct snd_soc_dapm_context *dapm) snd_soc_dapm_nc_pin(dapm, "LINEOUT"); /* At any time the WM9081 is active it will have this clock */ - return snd_soc_codec_set_sysclk(dapm->codec, WM9081_SYSCLK_MCLK, + return snd_soc_codec_set_sysclk(dapm->codec, WM9081_SYSCLK_MCLK, 0, 48000 * 256, 0); } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index fd173f6..187ddfd 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2670,7 +2670,7 @@ int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, if (dai->driver && dai->driver->ops->set_sysclk) return dai->driver->ops->set_sysclk(dai, clk_id, freq, dir); else if (dai->codec && dai->codec->driver->set_sysclk) - return dai->codec->driver->set_sysclk(dai->codec, clk_id, + return dai->codec->driver->set_sysclk(dai->codec, clk_id, 0, freq, dir); else return -EINVAL; @@ -2681,16 +2681,18 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk); * snd_soc_codec_set_sysclk - configure CODEC system or master clock. * @codec: CODEC * @clk_id: DAI specific clock ID + * @source: Source for the clock * @freq: new clock frequency in Hz * @dir: new clock direction - input/output. * * Configures the CODEC master (MCLK) or system (SYSCLK) clocking. */ int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, - unsigned int freq, int dir) + int source, unsigned int freq, int dir) { if (codec->driver->set_sysclk) - return codec->driver->set_sysclk(codec, clk_id, freq, dir); + return codec->driver->set_sysclk(codec, clk_id, source, + freq, dir); else return -EINVAL; } -- 1.7.5.4

3 3

[alsa-devel] [PATCH] ASoC: Remove unused mutex from WM9090 driver
by Mark Brown 31 Aug '11

31 Aug '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9090.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index 4de1220..f2f3077 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -139,7 +139,6 @@ static const u16 wm9090_reg_defaults[] = { /* This struct is used to save the context */ struct wm9090_priv { - struct mutex mutex; struct wm9090_platform_data pdata; void *control_data; }; @@ -663,7 +662,6 @@ static int wm9090_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm9090); wm9090->control_data = i2c; - mutex_init(&wm9090->mutex); ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm9090, NULL, 0); -- 1.7.5.4

2 1

[alsa-devel] [PATCH 1/3] ASoC: Add device tree binding for WM8770
by Mark Brown 31 Aug '11

31 Aug '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- Documentation/devicetree/bindings/sound/wm8770.txt | 16 ++++++++++++++++ sound/soc/codecs/wm8770.c | 8 ++++++++ 2 files changed, 24 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/wm8770.txt diff --git a/Documentation/devicetree/bindings/sound/wm8770.txt b/Documentation/devicetree/bindings/sound/wm8770.txt new file mode 100644 index 0000000..866e00c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8770.txt @@ -0,0 +1,16 @@ +WM8770 audio CODEC + +This device supports SPI. + +Required properties: + + - compatible : "wlf,wm8770" + + - reg : the chip select number. + +Example: + +codec: wm8770@1 { + compatible = "wlf,wm8770"; + reg = <1>; +}; diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c index 19b92ba..aa05e65 100644 --- a/sound/soc/codecs/wm8770.c +++ b/sound/soc/codecs/wm8770.c @@ -14,6 +14,7 @@ #include <linux/moduleparam.h> #include <linux/init.h> #include <linux/delay.h> +#include <linux/of_device.h> #include <linux/pm.h> #include <linux/platform_device.h> #include <linux/spi/spi.h> @@ -684,6 +685,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8770 = { .reg_cache_default = wm8770_reg_defs }; +static const struct of_device_id wm8770_of_match[] = { + { .compatible = "wlf,wm8770", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8770_of_match); + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8770_spi_probe(struct spi_device *spi) { @@ -715,6 +722,7 @@ static struct spi_driver wm8770_spi_driver = { .driver = { .name = "wm8770", .owner = THIS_MODULE, + .of_match_table = wm8770_of_match, }, .probe = wm8770_spi_probe, .remove = __devexit_p(wm8770_spi_remove) -- 1.7.5.4

5 8

[alsa-devel] [PATCH] ASoC: Remove redundant -codec from WM8776 driver name
by Mark Brown 31 Aug '11

31 Aug '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8776.c | 4 ++-- sound/soc/fsl/p1022_ds.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index 8e7953b..367a990 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -481,7 +481,7 @@ static int __devexit wm8776_spi_remove(struct spi_device *spi) static struct spi_driver wm8776_spi_driver = { .driver = { - .name = "wm8776-codec", + .name = "wm8776", .owner = THIS_MODULE, }, .probe = wm8776_spi_probe, @@ -525,7 +525,7 @@ MODULE_DEVICE_TABLE(i2c, wm8776_i2c_id); static struct i2c_driver wm8776_i2c_driver = { .driver = { - .name = "wm8776-codec", + .name = "wm8776", .owner = THIS_MODULE, }, .probe = wm8776_i2c_probe, diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index fcb862e..e8849ed 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -267,7 +267,7 @@ static int codec_node_dev_name(struct device_node *np, char *buf, size_t len) if (bus < 0) return bus; - snprintf(buf, len, "%s-codec.%u-%04x", temp, bus, addr); + snprintf(buf, len, "%s.%u-%04x", temp, bus, addr); return 0; } -- 1.7.5.4

3 2

[alsa-devel] [PATCH] ASoC: imx: Fix build warning of unused 'card' variable
by Fabio Estevam 31 Aug '11

31 Aug '11

Fixes the following warning: CC sound/soc/imx/imx-pcm-fiq.o sound/soc/imx/imx-pcm-fiq.c: In function 'imx_pcm_fiq_new': sound/soc/imx/imx-pcm-fiq.c:243: warning: unused variable 'card' CC sound/soc/imx/imx-pcm-dma-mx2.o Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/imx/imx-pcm-fiq.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c index 309c59e..7945625 100644 --- a/sound/soc/imx/imx-pcm-fiq.c +++ b/sound/soc/imx/imx-pcm-fiq.c @@ -240,7 +240,6 @@ static int ssi_irq = 0; static int imx_pcm_fiq_new(struct snd_soc_pcm_runtime *rtd) { - struct snd_card *card = rtd->card->snd_card; struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; int ret; -- 1.6.0.4

2 1

[alsa-devel] [PATCH 1/3] ASoC: snd_soc_codec_{readable, writable}_register change default to true
by Lars-Peter Clausen 31 Aug '11

31 Aug '11

Change the default return value of snd_soc_codec_{readable,writable}_register to true when no codec specific callback for this function is given. Otherwise all registers of that codec will neither be readable nor writable, which is most certainly not what we want. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-core.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index fd173f6..4065d4e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1635,7 +1635,7 @@ int snd_soc_codec_readable_register(struct snd_soc_codec *codec, if (codec->readable_register) return codec->readable_register(codec, reg); else - return 0; + return 1; } EXPORT_SYMBOL_GPL(snd_soc_codec_readable_register); @@ -1653,7 +1653,7 @@ int snd_soc_codec_writable_register(struct snd_soc_codec *codec, if (codec->writable_register) return codec->writable_register(codec, reg); else - return 0; + return 1; } EXPORT_SYMBOL_GPL(snd_soc_codec_writable_register); -- 1.7.2.5

3 4

[alsa-devel] [PATCH] ASoC: soc-dapm: Fix parameter comment for snd_soc_dapm_free
by Peter Ujfalusi 31 Aug '11

31 Aug '11

We have dapm_context instead of codec parameter. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hello Mark, Liam, Bumped to this while adding the snd_soc_dapm_attach_dai_supply. It must have been overlooked, when the API changed. Peter sound/soc/soc-dapm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8839202..5072b3b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2763,7 +2763,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_ignore_suspend); /** * snd_soc_dapm_free - free dapm resources - * @card: SoC device + * @dapm: DAPM context * * Free all dapm widgets and resources. */ -- 1.7.6.1

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Re: [alsa-devel] [PATCH] ASoC: Remove unused variable 'card' in imx_pcm_fiq_new
by Dong Aisheng-B29396 31 Aug '11

31 Aug '11

> -----Original Message----- > From: linux-kernel-owner(a)vger.kernel.org [mailto:linux-kernel- > owner(a)vger.kernel.org] On Behalf Of Axel Lin > Sent: Wednesday, August 31, 2011 1:50 PM > To: linux-kernel(a)vger.kernel.org > Cc: Sascha Hauer; Liam Girdwood; Mark Brown; alsa-devel(a)alsa-project.org > Subject: [PATCH] ASoC: Remove unused variable 'card' in imx_pcm_fiq_new > > Fix below build warning: It seems Fabio already sent this out a few days ago. Please refer to: http://comments.gmane.org/gmane.linux.alsa.devel/88686 > CC sound/soc/imx/imx-pcm-fiq.o > sound/soc/imx/imx-pcm-fiq.c: In function 'imx_pcm_fiq_new': > sound/soc/imx/imx-pcm-fiq.c:243: warning: unused variable 'card' > > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> > --- > sound/soc/imx/imx-pcm-fiq.c | 1 - > 1 files changed, 0 insertions(+), 1 deletions(-) > > diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c > index 309c59e..7945625 100644 > --- a/sound/soc/imx/imx-pcm-fiq.c > +++ b/sound/soc/imx/imx-pcm-fiq.c > @@ -240,7 +240,6 @@ static int ssi_irq = 0; > > static int imx_pcm_fiq_new(struct snd_soc_pcm_runtime *rtd) { > - struct snd_card *card = rtd->card->snd_card; > struct snd_soc_dai *dai = rtd->cpu_dai; > struct snd_pcm *pcm = rtd->pcm; > int ret; > -- > 1.7.4.1 > > > > -- > To unsubscribe from this list: send the line "unsubscribe linux-kernel" > in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/

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[alsa-devel] EeeTop ET1611PUT pin issue?
by Nicholas Hall 30 Aug '11

30 Aug '11

Greetings, Asus released a new line of EeeTop. I get audio working fine out the rear headphone jack. Although the internal speakers output nothing. I've been doing research and it seems quite possible that the hardware is pinned out unusually. Does this seem like the case? alsa-info http://www.alsa-project.org/db/?f=cf43f9508ed2fde7792245f0f760dd76787a7776 My other thought was that the hardware is not reporting the state of the headphone jack in use. Are there any pointers on identifying and fixing this issue? I tried Asus phone support (LOL!, what a joke). Do I need to get in contact with an Asus engineer, or does the alsa-info output have everything I need. I must admit, a lot of the output is foreign to me. Thanks in advanced! -- Nicholas Hall Software Developer GrandCare Systems

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[alsa-devel] [PATCH v4 1/1] ASoC: omap-mcpdm: Replace legacy driver
by Peter Ujfalusi 30 Aug '11

30 Aug '11

From: Misael Lopez Cruz <misael.lopez(a)ti.com> Reasons for the replacement: The current driver for McPDM was developed to support the legacy mode only. In preparation for the ABE support the current driver stack need the be replaced. The new driver is much simpler, easier to extend, and it also fixes some of the issues with the old stack. Main changes: - single file for omap-mcpdm (mcpdm.c/h removed) - Define names for registers, bits cleaned up, prefixed - Full-duplex audio operation (arecord | aplay) has been fixed - Less code McPDM need to be turned off after all streams has been stopped. This might cause pop noise on the output, if the codec's DAC is still powered at this time. Signed-off-by: Misael Lopez Cruz <misael.lopez(a)ti.com> Signed-off-by: Liam Girdwood <lrg(a)ti.com> Signed-off-by: Sebastien Guiriec <s-guiriec(a)ti.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi Mark, Liam, Changes since v3: - The delayed shutdown part has been taken out from the new driver for now. The McPDM interface is turned off at stream stop time, when all streams has been already stopped. This can create pop on the output until I can delay the power off after the codec's DAC power down, but before the full power down of the codec. Regards, Peter sound/soc/omap/Makefile | 2 +- sound/soc/omap/mcpdm.c | 474 -------------------------- sound/soc/omap/mcpdm.h | 152 --------- sound/soc/omap/omap-mcpdm.c | 789 ++++++++++++++++++++++++++++--------------- sound/soc/omap/omap-mcpdm.h | 95 ++++++ sound/soc/omap/sdp4430.c | 2 +- 6 files changed, 607 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 (44%) create mode 100644 sound/soc/omap/omap-mcpdm.h diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 59e2c8d..052fd75 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -1,7 +1,7 @@ # OMAP Platform Support snd-soc-omap-objs := omap-pcm.o snd-soc-omap-mcbsp-objs := omap-mcbsp.o -snd-soc-omap-mcpdm-objs := omap-mcpdm.o mcpdm.o +snd-soc-omap-mcpdm-objs := omap-mcpdm.o snd-soc-omap-hdmi-objs := omap-hdmi.o obj-$(CONFIG_SND_OMAP_SOC) += snd-soc-omap.o diff --git a/sound/soc/omap/mcpdm.c b/sound/soc/omap/mcpdm.c deleted file mode 100644 index d29cc98..0000000 --- a/sound/soc/omap/mcpdm.c +++ /dev/null @@ -1,474 +0,0 @@ -/* - * mcpdm.c -- McPDM interface driver - * - * Author: Jorge Eduardo Candelaria <x0107209(a)ti.com> - * Copyright (C) 2009 - Texas Instruments, 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/module.h> -#include <linux/init.h> -#include <linux/device.h> -#include <linux/platform_device.h> -#include <linux/wait.h> -#include <linux/slab.h> -#include <linux/interrupt.h> -#include <linux/err.h> -#include <linux/pm_runtime.h> -#include <linux/delay.h> -#include <linux/io.h> -#include <linux/irq.h> - -#include "mcpdm.h" - -static struct omap_mcpdm *mcpdm; - -static inline void omap_mcpdm_write(u16 reg, u32 val) -{ - __raw_writel(val, mcpdm->io_base + reg); -} - -static inline int omap_mcpdm_read(u16 reg) -{ - return __raw_readl(mcpdm->io_base + reg); -} - -static void omap_mcpdm_reg_dump(void) -{ - dev_dbg(mcpdm->dev, "***********************\n"); - dev_dbg(mcpdm->dev, "IRQSTATUS_RAW: 0x%04x\n", - omap_mcpdm_read(MCPDM_IRQSTATUS_RAW)); - dev_dbg(mcpdm->dev, "IRQSTATUS: 0x%04x\n", - omap_mcpdm_read(MCPDM_IRQSTATUS)); - dev_dbg(mcpdm->dev, "IRQENABLE_SET: 0x%04x\n", - omap_mcpdm_read(MCPDM_IRQENABLE_SET)); - dev_dbg(mcpdm->dev, "IRQENABLE_CLR: 0x%04x\n", - omap_mcpdm_read(MCPDM_IRQENABLE_CLR)); - dev_dbg(mcpdm->dev, "IRQWAKE_EN: 0x%04x\n", - omap_mcpdm_read(MCPDM_IRQWAKE_EN)); - dev_dbg(mcpdm->dev, "DMAENABLE_SET: 0x%04x\n", - omap_mcpdm_read(MCPDM_DMAENABLE_SET)); - dev_dbg(mcpdm->dev, "DMAENABLE_CLR: 0x%04x\n", - omap_mcpdm_read(MCPDM_DMAENABLE_CLR)); - dev_dbg(mcpdm->dev, "DMAWAKEEN: 0x%04x\n", - omap_mcpdm_read(MCPDM_DMAWAKEEN)); - dev_dbg(mcpdm->dev, "CTRL: 0x%04x\n", - omap_mcpdm_read(MCPDM_CTRL)); - dev_dbg(mcpdm->dev, "DN_DATA: 0x%04x\n", - omap_mcpdm_read(MCPDM_DN_DATA)); - dev_dbg(mcpdm->dev, "UP_DATA: 0x%04x\n", - omap_mcpdm_read(MCPDM_UP_DATA)); - dev_dbg(mcpdm->dev, "FIFO_CTRL_DN: 0x%04x\n", - omap_mcpdm_read(MCPDM_FIFO_CTRL_DN)); - dev_dbg(mcpdm->dev, "FIFO_CTRL_UP: 0x%04x\n", - omap_mcpdm_read(MCPDM_FIFO_CTRL_UP)); - dev_dbg(mcpdm->dev, "DN_OFFSET: 0x%04x\n", - omap_mcpdm_read(MCPDM_DN_OFFSET)); - dev_dbg(mcpdm->dev, "***********************\n"); -} - -/* - * Takes the McPDM module in and out of reset state. - * Uplink and downlink can be reset individually. - */ -static void omap_mcpdm_reset_capture(int reset) -{ - int ctrl = omap_mcpdm_read(MCPDM_CTRL); - - if (reset) - ctrl |= SW_UP_RST; - else - ctrl &= ~SW_UP_RST; - - omap_mcpdm_write(MCPDM_CTRL, ctrl); -} - -static void omap_mcpdm_reset_playback(int reset) -{ - int ctrl = omap_mcpdm_read(MCPDM_CTRL); - - if (reset) - ctrl |= SW_DN_RST; - else - ctrl &= ~SW_DN_RST; - - omap_mcpdm_write(MCPDM_CTRL, ctrl); -} - -/* - * Enables the transfer through the PDM interface to/from the Phoenix - * codec by enabling the corresponding UP or DN channels. - */ -void omap_mcpdm_start(int stream) -{ - int ctrl = omap_mcpdm_read(MCPDM_CTRL); - - if (stream) - ctrl |= mcpdm->up_channels; - else - ctrl |= mcpdm->dn_channels; - - omap_mcpdm_write(MCPDM_CTRL, ctrl); -} - -/* - * Disables the transfer through the PDM interface to/from the Phoenix - * codec by disabling the corresponding UP or DN channels. - */ -void omap_mcpdm_stop(int stream) -{ - int ctrl = omap_mcpdm_read(MCPDM_CTRL); - - if (stream) - ctrl &= ~mcpdm->up_channels; - else - ctrl &= ~mcpdm->dn_channels; - - omap_mcpdm_write(MCPDM_CTRL, ctrl); -} - -/* - * Configures McPDM uplink for audio recording. - * This function should be called before omap_mcpdm_start. - */ -int omap_mcpdm_capture_open(struct omap_mcpdm_link *uplink) -{ - int irq_mask = 0; - int ctrl; - - if (!uplink) - return -EINVAL; - - mcpdm->uplink = uplink; - - /* Enable irq request generation */ - irq_mask |= uplink->irq_mask & MCPDM_UPLINK_IRQ_MASK; - omap_mcpdm_write(MCPDM_IRQENABLE_SET, irq_mask); - - /* Configure uplink threshold */ - if (uplink->threshold > UP_THRES_MAX) - uplink->threshold = UP_THRES_MAX; - - omap_mcpdm_write(MCPDM_FIFO_CTRL_UP, uplink->threshold); - - /* Configure DMA controller */ - omap_mcpdm_write(MCPDM_DMAENABLE_SET, DMA_UP_ENABLE); - - /* Set pdm out format */ - ctrl = omap_mcpdm_read(MCPDM_CTRL); - ctrl &= ~PDMOUTFORMAT; - ctrl |= uplink->format & PDMOUTFORMAT; - - /* Uplink channels */ - mcpdm->up_channels = uplink->channels & (PDM_UP_MASK | PDM_STATUS_MASK); - - omap_mcpdm_write(MCPDM_CTRL, ctrl); - - return 0; -} - -/* - * Configures McPDM downlink for audio playback. - * This function should be called before omap_mcpdm_start. - */ -int omap_mcpdm_playback_open(struct omap_mcpdm_link *downlink) -{ - int irq_mask = 0; - int ctrl; - - if (!downlink) - return -EINVAL; - - mcpdm->downlink = downlink; - - /* Enable irq request generation */ - irq_mask |= downlink->irq_mask & MCPDM_DOWNLINK_IRQ_MASK; - omap_mcpdm_write(MCPDM_IRQENABLE_SET, irq_mask); - - /* Configure uplink threshold */ - if (downlink->threshold > DN_THRES_MAX) - downlink->threshold = DN_THRES_MAX; - - omap_mcpdm_write(MCPDM_FIFO_CTRL_DN, downlink->threshold); - - /* Enable DMA request generation */ - omap_mcpdm_write(MCPDM_DMAENABLE_SET, DMA_DN_ENABLE); - - /* Set pdm out format */ - ctrl = omap_mcpdm_read(MCPDM_CTRL); - ctrl &= ~PDMOUTFORMAT; - ctrl |= downlink->format & PDMOUTFORMAT; - - /* Downlink channels */ - mcpdm->dn_channels = downlink->channels & (PDM_DN_MASK | PDM_CMD_MASK); - - omap_mcpdm_write(MCPDM_CTRL, ctrl); - - return 0; -} - -/* - * Cleans McPDM uplink configuration. - * This function should be called when the stream is closed. - */ -int omap_mcpdm_capture_close(struct omap_mcpdm_link *uplink) -{ - int irq_mask = 0; - - if (!uplink) - return -EINVAL; - - /* Disable irq request generation */ - irq_mask |= uplink->irq_mask & MCPDM_UPLINK_IRQ_MASK; - omap_mcpdm_write(MCPDM_IRQENABLE_CLR, irq_mask); - - /* Disable DMA request generation */ - omap_mcpdm_write(MCPDM_DMAENABLE_CLR, DMA_UP_ENABLE); - - /* Clear Downlink channels */ - mcpdm->up_channels = 0; - - mcpdm->uplink = NULL; - - return 0; -} - -/* - * Cleans McPDM downlink configuration. - * This function should be called when the stream is closed. - */ -int omap_mcpdm_playback_close(struct omap_mcpdm_link *downlink) -{ - int irq_mask = 0; - - if (!downlink) - return -EINVAL; - - /* Disable irq request generation */ - irq_mask |= downlink->irq_mask & MCPDM_DOWNLINK_IRQ_MASK; - omap_mcpdm_write(MCPDM_IRQENABLE_CLR, irq_mask); - - /* Disable DMA request generation */ - omap_mcpdm_write(MCPDM_DMAENABLE_CLR, DMA_DN_ENABLE); - - /* clear Downlink channels */ - mcpdm->dn_channels = 0; - - mcpdm->downlink = NULL; - - return 0; -} - -static irqreturn_t omap_mcpdm_irq_handler(int irq, void *dev_id) -{ - struct omap_mcpdm *mcpdm_irq = dev_id; - int irq_status; - - irq_status = omap_mcpdm_read(MCPDM_IRQSTATUS); - - /* Acknowledge irq event */ - omap_mcpdm_write(MCPDM_IRQSTATUS, irq_status); - - if (irq & MCPDM_DN_IRQ_FULL) { - dev_err(mcpdm_irq->dev, "DN FIFO error %x\n", irq_status); - omap_mcpdm_reset_playback(1); - omap_mcpdm_playback_open(mcpdm_irq->downlink); - omap_mcpdm_reset_playback(0); - } - - if (irq & MCPDM_DN_IRQ_EMPTY) { - dev_err(mcpdm_irq->dev, "DN FIFO error %x\n", irq_status); - omap_mcpdm_reset_playback(1); - omap_mcpdm_playback_open(mcpdm_irq->downlink); - omap_mcpdm_reset_playback(0); - } - - if (irq & MCPDM_DN_IRQ) { - dev_dbg(mcpdm_irq->dev, "DN write request\n"); - } - - if (irq & MCPDM_UP_IRQ_FULL) { - dev_err(mcpdm_irq->dev, "UP FIFO error %x\n", irq_status); - omap_mcpdm_reset_capture(1); - omap_mcpdm_capture_open(mcpdm_irq->uplink); - omap_mcpdm_reset_capture(0); - } - - if (irq & MCPDM_UP_IRQ_EMPTY) { - dev_err(mcpdm_irq->dev, "UP FIFO error %x\n", irq_status); - omap_mcpdm_reset_capture(1); - omap_mcpdm_capture_open(mcpdm_irq->uplink); - omap_mcpdm_reset_capture(0); - } - - if (irq & MCPDM_UP_IRQ) { - dev_dbg(mcpdm_irq->dev, "UP write request\n"); - } - - return IRQ_HANDLED; -} - - int omap_mcpdm_request(void) -{ - int ret; - - pm_runtime_get_sync(mcpdm->dev); - - spin_lock(&mcpdm->lock); - - if (!mcpdm->free) { - dev_err(mcpdm->dev, "McPDM interface is in use\n"); - spin_unlock(&mcpdm->lock); - ret = -EBUSY; - goto err; - } - mcpdm->free = 0; - - spin_unlock(&mcpdm->lock); - - /* Disable lines while request is ongoing */ - omap_mcpdm_write(MCPDM_CTRL, 0x00); - - ret = request_irq(mcpdm->irq, omap_mcpdm_irq_handler, - 0, "McPDM", (void *)mcpdm); - if (ret) { - dev_err(mcpdm->dev, "Request for McPDM IRQ failed\n"); - goto err; - } - - return 0; - -err: - mcpdm->free = 1; - pm_runtime_put_sync(mcpdm->dev); - return ret; -} - -void omap_mcpdm_free(void) -{ - spin_lock(&mcpdm->lock); - if (mcpdm->free) { - dev_err(mcpdm->dev, "McPDM interface is already free\n"); - spin_unlock(&mcpdm->lock); - return; - } - mcpdm->free = 1; - spin_unlock(&mcpdm->lock); - - pm_runtime_put_sync(mcpdm->dev); - - free_irq(mcpdm->irq, (void *)mcpdm); -} - -/* Enable/disable DC offset cancelation for the analog - * headset path (PDM channels 1 and 2). - */ -int omap_mcpdm_set_offset(int offset1, int offset2) -{ - int offset; - - if ((offset1 > DN_OFST_MAX) || (offset2 > DN_OFST_MAX)) - return -EINVAL; - - offset = (offset1 << DN_OFST_RX1) | (offset2 << DN_OFST_RX2); - - /* offset cancellation for channel 1 */ - if (offset1) - offset |= DN_OFST_RX1_EN; - else - offset &= ~DN_OFST_RX1_EN; - - /* offset cancellation for channel 2 */ - if (offset2) - offset |= DN_OFST_RX2_EN; - else - offset &= ~DN_OFST_RX2_EN; - - omap_mcpdm_write(MCPDM_DN_OFFSET, offset); - - return 0; -} - -int __devinit omap_mcpdm_probe(struct platform_device *pdev) -{ - struct resource *res; - int ret = 0; - - mcpdm = kzalloc(sizeof(struct omap_mcpdm), GFP_KERNEL); - if (!mcpdm) { - ret = -ENOMEM; - goto exit; - } - - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (res == NULL) { - dev_err(&pdev->dev, "no resource\n"); - goto err_resource; - } - - spin_lock_init(&mcpdm->lock); - mcpdm->free = 1; - - if (!request_mem_region(res->start, resource_size(res), "McPDM")) { - ret = -EBUSY; - goto err_resource; - } - - mcpdm->io_base = ioremap(res->start, resource_size(res)); - if (!mcpdm->io_base) { - ret = -ENOMEM; - goto err_remap; - } - - mcpdm->irq = platform_get_irq(pdev, 0); - - mcpdm->dev = &pdev->dev; - platform_set_drvdata(pdev, mcpdm); - - pm_runtime_enable(mcpdm->dev); - - return 0; - -err_remap: - release_mem_region(res->start, resource_size(res)); -err_resource: - kfree(mcpdm); -exit: - return ret; -} - -int __devexit omap_mcpdm_remove(struct platform_device *pdev) -{ - struct omap_mcpdm *mcpdm_ptr = platform_get_drvdata(pdev); - struct resource *res; - - platform_set_drvdata(pdev, NULL); - - pm_runtime_disable(mcpdm_ptr->dev); - - iounmap(mcpdm_ptr->io_base); - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - release_mem_region(res->start, resource_size(res)); - - mcpdm_ptr->free = 0; - mcpdm_ptr->dev = NULL; - - kfree(mcpdm_ptr); - - return 0; -} - diff --git a/sound/soc/omap/mcpdm.h b/sound/soc/omap/mcpdm.h deleted file mode 100644 index b055ad1..0000000 --- a/sound/soc/omap/mcpdm.h +++ /dev/null @@ -1,152 +0,0 @@ -/* - * mcpdm.h -- Defines for McPDM driver - * - * Author: Jorge Eduardo Candelaria <x0107209(a)ti.com> - * - * 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 - * - */ - -/* McPDM registers */ - -#define MCPDM_REVISION 0x00 -#define MCPDM_SYSCONFIG 0x10 -#define MCPDM_IRQSTATUS_RAW 0x24 -#define MCPDM_IRQSTATUS 0x28 -#define MCPDM_IRQENABLE_SET 0x2C -#define MCPDM_IRQENABLE_CLR 0x30 -#define MCPDM_IRQWAKE_EN 0x34 -#define MCPDM_DMAENABLE_SET 0x38 -#define MCPDM_DMAENABLE_CLR 0x3C -#define MCPDM_DMAWAKEEN 0x40 -#define MCPDM_CTRL 0x44 -#define MCPDM_DN_DATA 0x48 -#define MCPDM_UP_DATA 0x4C -#define MCPDM_FIFO_CTRL_DN 0x50 -#define MCPDM_FIFO_CTRL_UP 0x54 -#define MCPDM_DN_OFFSET 0x58 - -/* - * MCPDM_IRQ bit fields - * IRQSTATUS_RAW, IRQSTATUS, IRQENABLE_SET, IRQENABLE_CLR - */ - -#define MCPDM_DN_IRQ (1 << 0) -#define MCPDM_DN_IRQ_EMPTY (1 << 1) -#define MCPDM_DN_IRQ_ALMST_EMPTY (1 << 2) -#define MCPDM_DN_IRQ_FULL (1 << 3) - -#define MCPDM_UP_IRQ (1 << 8) -#define MCPDM_UP_IRQ_EMPTY (1 << 9) -#define MCPDM_UP_IRQ_ALMST_FULL (1 << 10) -#define MCPDM_UP_IRQ_FULL (1 << 11) - -#define MCPDM_DOWNLINK_IRQ_MASK 0x00F -#define MCPDM_UPLINK_IRQ_MASK 0xF00 - -/* - * MCPDM_DMAENABLE bit fields - */ - -#define DMA_DN_ENABLE 0x1 -#define DMA_UP_ENABLE 0x2 - -/* - * MCPDM_CTRL bit fields - */ - -#define PDM_UP1_EN 0x0001 -#define PDM_UP2_EN 0x0002 -#define PDM_UP3_EN 0x0004 -#define PDM_DN1_EN 0x0008 -#define PDM_DN2_EN 0x0010 -#define PDM_DN3_EN 0x0020 -#define PDM_DN4_EN 0x0040 -#define PDM_DN5_EN 0x0080 -#define PDMOUTFORMAT 0x0100 -#define CMD_INT 0x0200 -#define STATUS_INT 0x0400 -#define SW_UP_RST 0x0800 -#define SW_DN_RST 0x1000 -#define PDM_UP_MASK 0x007 -#define PDM_DN_MASK 0x0F8 -#define PDM_CMD_MASK 0x200 -#define PDM_STATUS_MASK 0x400 - - -#define PDMOUTFORMAT_LJUST (0 << 8) -#define PDMOUTFORMAT_RJUST (1 << 8) - -/* - * MCPDM_FIFO_CTRL bit fields - */ - -#define UP_THRES_MAX 0xF -#define DN_THRES_MAX 0xF - -/* - * MCPDM_DN_OFFSET bit fields - */ - -#define DN_OFST_RX1_EN 0x0001 -#define DN_OFST_RX2_EN 0x0100 - -#define DN_OFST_RX1 1 -#define DN_OFST_RX2 9 -#define DN_OFST_MAX 0x1F - -#define MCPDM_UPLINK 1 -#define MCPDM_DOWNLINK 2 - -struct omap_mcpdm_link { - int irq_mask; - int threshold; - int format; - int channels; -}; - -struct omap_mcpdm_platform_data { - unsigned long phys_base; - u16 irq; -}; - -struct omap_mcpdm { - struct device *dev; - unsigned long phys_base; - void __iomem *io_base; - u8 free; - int irq; - - spinlock_t lock; - struct omap_mcpdm_platform_data *pdata; - struct omap_mcpdm_link *downlink; - struct omap_mcpdm_link *uplink; - struct completion irq_completion; - - int dn_channels; - int up_channels; -}; - -extern void omap_mcpdm_start(int stream); -extern void omap_mcpdm_stop(int stream); -extern int omap_mcpdm_capture_open(struct omap_mcpdm_link *uplink); -extern int omap_mcpdm_playback_open(struct omap_mcpdm_link *downlink); -extern int omap_mcpdm_capture_close(struct omap_mcpdm_link *uplink); -extern int omap_mcpdm_playback_close(struct omap_mcpdm_link *downlink); -extern int omap_mcpdm_request(void); -extern void omap_mcpdm_free(void); -extern int omap_mcpdm_set_offset(int offset1, int offset2); -int __devinit omap_mcpdm_probe(struct platform_device *pdev); -int __devexit omap_mcpdm_remove(struct platform_device *pdev); diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c dissimilarity index 44% index 0820b9e..159a5e9 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -1,279 +1,510 @@ -/* - * omap-mcpdm.c -- OMAP ALSA SoC DAI driver using McPDM port - * - * Copyright (C) 2009 Texas Instruments - * - * Author: Misael Lopez Cruz <x0052729(a)ti.com> - * Contact: Jorge Eduardo Candelaria <x0107209(a)ti.com> - * Margarita Olaya <magi.olaya(a)ti.com> - * - * 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/init.h> -#include <linux/module.h> -#include <linux/device.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/initval.h> -#include <sound/soc.h> - -#include <plat/dma.h> -#include <plat/mcbsp.h> -#include "mcpdm.h" -#include "omap-pcm.h" - -struct omap_mcpdm_data { - struct omap_mcpdm_link *links; - int active; -}; - -static struct omap_mcpdm_link omap_mcpdm_links[] = { - /* downlink */ - { - .irq_mask = MCPDM_DN_IRQ_EMPTY | MCPDM_DN_IRQ_FULL, - .threshold = 2, - .format = PDMOUTFORMAT_LJUST, - }, - /* uplink */ - { - .irq_mask = MCPDM_UP_IRQ_EMPTY | MCPDM_UP_IRQ_FULL, - .threshold = UP_THRES_MAX - 3, - .format = PDMOUTFORMAT_LJUST, - }, -}; - -static struct omap_mcpdm_data mcpdm_data = { - .links = omap_mcpdm_links, - .active = 0, -}; - -/* - * Stream DMA parameters - */ -static struct omap_pcm_dma_data omap_mcpdm_dai_dma_params[] = { - { - .name = "Audio playback", - .dma_req = OMAP44XX_DMA_MCPDM_DL, - .data_type = OMAP_DMA_DATA_TYPE_S32, - .sync_mode = OMAP_DMA_SYNC_PACKET, - .packet_size = 16, - .port_addr = OMAP44XX_MCPDM_L3_BASE + MCPDM_DN_DATA, - }, - { - .name = "Audio capture", - .dma_req = OMAP44XX_DMA_MCPDM_UP, - .data_type = OMAP_DMA_DATA_TYPE_S32, - .sync_mode = OMAP_DMA_SYNC_PACKET, - .packet_size = 16, - .port_addr = OMAP44XX_MCPDM_L3_BASE + MCPDM_UP_DATA, - }, -}; - -static int omap_mcpdm_dai_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - int err = 0; - - if (!dai->active) - err = omap_mcpdm_request(); - - return err; -} - -static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - if (!dai->active) - omap_mcpdm_free(); -} - -static int omap_mcpdm_dai_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) -{ - struct omap_mcpdm_data *mcpdm_priv = snd_soc_dai_get_drvdata(dai); - int stream = substream->stream; - int err = 0; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (!mcpdm_priv->active++) - omap_mcpdm_start(stream); - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (!--mcpdm_priv->active) - omap_mcpdm_stop(stream); - break; - default: - err = -EINVAL; - } - - return err; -} - -static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct omap_mcpdm_data *mcpdm_priv = snd_soc_dai_get_drvdata(dai); - struct omap_mcpdm_link *mcpdm_links = mcpdm_priv->links; - struct omap_pcm_dma_data *dma_data; - int threshold; - int stream = substream->stream; - int channels, err, link_mask = 0; - - channels = params_channels(params); - switch (channels) { - case 4: - if (stream == SNDRV_PCM_STREAM_CAPTURE) - /* up to 2 channels for capture */ - return -EINVAL; - link_mask |= 1 << 3; - case 3: - if (stream == SNDRV_PCM_STREAM_CAPTURE) - /* up to 2 channels for capture */ - return -EINVAL; - link_mask |= 1 << 2; - case 2: - link_mask |= 1 << 1; - case 1: - link_mask |= 1 << 0; - break; - default: - /* unsupported number of channels */ - return -EINVAL; - } - - dma_data = &omap_mcpdm_dai_dma_params[stream]; - threshold = mcpdm_links[stream].threshold; - - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - mcpdm_links[stream].channels = link_mask << 3; - dma_data->packet_size = (DN_THRES_MAX - threshold) * channels; - - err = omap_mcpdm_playback_open(&mcpdm_links[stream]); - } else { - mcpdm_links[stream].channels = link_mask << 0; - dma_data->packet_size = threshold * channels; - - err = omap_mcpdm_capture_open(&mcpdm_links[stream]); - } - - snd_soc_dai_set_dma_data(dai, substream, dma_data); - return err; -} - -static int omap_mcpdm_dai_hw_free(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct omap_mcpdm_data *mcpdm_priv = snd_soc_dai_get_drvdata(dai); - struct omap_mcpdm_link *mcpdm_links = mcpdm_priv->links; - int stream = substream->stream; - int err; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - err = omap_mcpdm_playback_close(&mcpdm_links[stream]); - else - err = omap_mcpdm_capture_close(&mcpdm_links[stream]); - - return err; -} - -static struct snd_soc_dai_ops omap_mcpdm_dai_ops = { - .startup = omap_mcpdm_dai_startup, - .shutdown = omap_mcpdm_dai_shutdown, - .trigger = omap_mcpdm_dai_trigger, - .hw_params = omap_mcpdm_dai_hw_params, - .hw_free = omap_mcpdm_dai_hw_free, -}; - -#define OMAP_MCPDM_RATES (SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) -#define OMAP_MCPDM_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) - -static int omap_mcpdm_dai_probe(struct snd_soc_dai *dai) -{ - snd_soc_dai_set_drvdata(dai, &mcpdm_data); - return 0; -} - -static struct snd_soc_dai_driver omap_mcpdm_dai = { - .probe = omap_mcpdm_dai_probe, - .playback = { - .channels_min = 1, - .channels_max = 4, - .rates = OMAP_MCPDM_RATES, - .formats = OMAP_MCPDM_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = OMAP_MCPDM_RATES, - .formats = OMAP_MCPDM_FORMATS, - }, - .ops = &omap_mcpdm_dai_ops, -}; - -static __devinit int asoc_mcpdm_probe(struct platform_device *pdev) -{ - int ret; - - ret = omap_mcpdm_probe(pdev); - if (ret < 0) - return ret; - ret = snd_soc_register_dai(&pdev->dev, &omap_mcpdm_dai); - if (ret < 0) - omap_mcpdm_remove(pdev); - return ret; -} - -static int __devexit asoc_mcpdm_remove(struct platform_device *pdev) -{ - snd_soc_unregister_dai(&pdev->dev); - omap_mcpdm_remove(pdev); - return 0; -} - -static struct platform_driver asoc_mcpdm_driver = { - .driver = { - .name = "omap-mcpdm", - .owner = THIS_MODULE, - }, - - .probe = asoc_mcpdm_probe, - .remove = __devexit_p(asoc_mcpdm_remove), -}; - -static int __init snd_omap_mcpdm_init(void) -{ - return platform_driver_register(&asoc_mcpdm_driver); -} -module_init(snd_omap_mcpdm_init); - -static void __exit snd_omap_mcpdm_exit(void) -{ - platform_driver_unregister(&asoc_mcpdm_driver); -} -module_exit(snd_omap_mcpdm_exit); - -MODULE_AUTHOR("Misael Lopez Cruz <x0052729(a)ti.com>"); -MODULE_DESCRIPTION("OMAP PDM SoC Interface"); -MODULE_LICENSE("GPL"); +/* + * omap-mcpdm.c -- OMAP ALSA SoC DAI driver using McPDM port + * + * Copyright (C) 2009 - 2011 Texas Instruments + * + * Author: Misael Lopez Cruz <misael.lopez(a)ti.com> + * Contact: Jorge Eduardo Candelaria <x0107209(a)ti.com> + * Margarita Olaya <magi.olaya(a)ti.com> + * Peter Ujfalusi <peter.ujfalusi(a)ti.com> + * + * 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/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/interrupt.h> +#include <linux/err.h> +#include <linux/io.h> +#include <linux/irq.h> +#include <linux/slab.h> +#include <linux/pm_runtime.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <plat/dma.h> +#include <plat/omap_hwmod.h> +#include "omap-mcpdm.h" +#include "omap-pcm.h" + +struct omap_mcpdm { + struct device *dev; + unsigned long phys_base; + void __iomem *io_base; + int irq; + + struct mutex mutex; + + /* channel data */ + u32 dn_channels; + u32 up_channels; + + /* McPDM FIFO thresholds */ + u32 dn_threshold; + u32 up_threshold; +}; + +/* + * Stream DMA parameters + */ +static struct omap_pcm_dma_data omap_mcpdm_dai_dma_params[] = { + { + .name = "Audio playback", + .dma_req = OMAP44XX_DMA_MCPDM_DL, + .data_type = OMAP_DMA_DATA_TYPE_S32, + .sync_mode = OMAP_DMA_SYNC_PACKET, + .port_addr = OMAP44XX_MCPDM_L3_BASE + MCPDM_REG_DN_DATA, + }, + { + .name = "Audio capture", + .dma_req = OMAP44XX_DMA_MCPDM_UP, + .data_type = OMAP_DMA_DATA_TYPE_S32, + .sync_mode = OMAP_DMA_SYNC_PACKET, + .port_addr = OMAP44XX_MCPDM_L3_BASE + MCPDM_REG_UP_DATA, + }, +}; + +static inline void omap_mcpdm_write(struct omap_mcpdm *mcpdm, u16 reg, u32 val) +{ + __raw_writel(val, mcpdm->io_base + reg); +} + +static inline int omap_mcpdm_read(struct omap_mcpdm *mcpdm, u16 reg) +{ + return __raw_readl(mcpdm->io_base + reg); +} + +#ifdef DEBUG +static void omap_mcpdm_reg_dump(struct omap_mcpdm *mcpdm) +{ + dev_dbg(mcpdm->dev, "***********************\n"); + dev_dbg(mcpdm->dev, "IRQSTATUS_RAW: 0x%04x\n", + omap_mcpdm_read(mcpdm, MCPDM_REG_IRQSTATUS_RAW)); + dev_dbg(mcpdm->dev, "IRQSTATUS: 0x%04x\n", + omap_mcpdm_read(mcpdm, MCPDM_REG_IRQSTATUS)); + dev_dbg(mcpdm->dev, "IRQENABLE_SET: 0x%04x\n", + omap_mcpdm_read(mcpdm, MCPDM_REG_IRQENABLE_SET)); + dev_dbg(mcpdm->dev, "IRQENABLE_CLR: 0x%04x\n", + omap_mcpdm_read(mcpdm, MCPDM_REG_IRQENABLE_CLR)); + dev_dbg(mcpdm->dev, "IRQWAKE_EN: 0x%04x\n", + omap_mcpdm_read(mcpdm, MCPDM_REG_IRQWAKE_EN)); + dev_dbg(mcpdm->dev, "DMAENABLE_SET: 0x%04x\n", + omap_mcpdm_read(mcpdm, MCPDM_REG_DMAENABLE_SET)); + dev_dbg(mcpdm->dev, "DMAENABLE_CLR: 0x%04x\n", + omap_mcpdm_read(mcpdm, MCPDM_REG_DMAENABLE_CLR)); + dev_dbg(mcpdm->dev, "DMAWAKEEN: 0x%04x\n", + omap_mcpdm_read(mcpdm, MCPDM_REG_DMAWAKEEN)); + dev_dbg(mcpdm->dev, "CTRL: 0x%04x\n", + omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL)); + dev_dbg(mcpdm->dev, "DN_DATA: 0x%04x\n", + omap_mcpdm_read(mcpdm, MCPDM_REG_DN_DATA)); + dev_dbg(mcpdm->dev, "UP_DATA: 0x%04x\n", + omap_mcpdm_read(mcpdm, MCPDM_REG_UP_DATA)); + dev_dbg(mcpdm->dev, "FIFO_CTRL_DN: 0x%04x\n", + omap_mcpdm_read(mcpdm, MCPDM_REG_FIFO_CTRL_DN)); + dev_dbg(mcpdm->dev, "FIFO_CTRL_UP: 0x%04x\n", + omap_mcpdm_read(mcpdm, MCPDM_REG_FIFO_CTRL_UP)); + dev_dbg(mcpdm->dev, "***********************\n"); +} +#else +static void omap_mcpdm_reg_dump(struct omap_mcpdm *mcpdm) {} +#endif + +/* + * Enables the transfer through the PDM interface to/from the Phoenix + * codec by enabling the corresponding UP or DN channels. + */ +static void omap_mcpdm_start(struct omap_mcpdm *mcpdm) +{ + u32 ctrl = omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL); + + ctrl |= (MCPDM_SW_DN_RST | MCPDM_SW_UP_RST); + omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl); + + ctrl |= mcpdm->dn_channels | mcpdm->up_channels; + omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl); + + ctrl &= ~(MCPDM_SW_DN_RST | MCPDM_SW_UP_RST); + omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl); +} + +/* + * Disables the transfer through the PDM interface to/from the Phoenix + * codec by disabling the corresponding UP or DN channels. + */ +static void omap_mcpdm_stop(struct omap_mcpdm *mcpdm) +{ + u32 ctrl = omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL); + + ctrl |= (MCPDM_SW_DN_RST | MCPDM_SW_UP_RST); + omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl); + + ctrl &= ~(mcpdm->dn_channels | mcpdm->up_channels); + omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl); + + ctrl &= ~(MCPDM_SW_DN_RST | MCPDM_SW_UP_RST); + omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl); + +} + +/* + * Is the physical McPDM interface active. + */ +static inline int omap_mcpdm_active(struct omap_mcpdm *mcpdm) +{ + return omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL) & + (MCPDM_PDM_DN_MASK | MCPDM_PDM_UP_MASK); +} + +/* + * Configures McPDM uplink, and downlink for audio. + * This function should be called before omap_mcpdm_start. + */ +static void omap_mcpdm_open_streams(struct omap_mcpdm *mcpdm) +{ + omap_mcpdm_write(mcpdm, MCPDM_REG_IRQENABLE_SET, + MCPDM_DN_IRQ_EMPTY | MCPDM_DN_IRQ_FULL | + MCPDM_UP_IRQ_EMPTY | MCPDM_UP_IRQ_FULL); + + omap_mcpdm_write(mcpdm, MCPDM_REG_FIFO_CTRL_DN, mcpdm->dn_threshold); + omap_mcpdm_write(mcpdm, MCPDM_REG_FIFO_CTRL_UP, mcpdm->up_threshold); + + omap_mcpdm_write(mcpdm, MCPDM_REG_DMAENABLE_SET, + MCPDM_DMA_DN_ENABLE | MCPDM_DMA_UP_ENABLE); +} + +/* + * Cleans McPDM uplink, and downlink configuration. + * This function should be called when the stream is closed. + */ +static void omap_mcpdm_close_streams(struct omap_mcpdm *mcpdm) +{ + /* Disable irq request generation for downlink */ + omap_mcpdm_write(mcpdm, MCPDM_REG_IRQENABLE_CLR, + MCPDM_DN_IRQ_EMPTY | MCPDM_DN_IRQ_FULL); + + /* Disable DMA request generation for downlink */ + omap_mcpdm_write(mcpdm, MCPDM_REG_DMAENABLE_CLR, MCPDM_DMA_DN_ENABLE); + + /* Disable irq request generation for uplink */ + omap_mcpdm_write(mcpdm, MCPDM_REG_IRQENABLE_CLR, + MCPDM_UP_IRQ_EMPTY | MCPDM_UP_IRQ_FULL); + + /* Disable DMA request generation for uplink */ + omap_mcpdm_write(mcpdm, MCPDM_REG_DMAENABLE_CLR, MCPDM_DMA_UP_ENABLE); +} + +static irqreturn_t omap_mcpdm_irq_handler(int irq, void *dev_id) +{ + struct omap_mcpdm *mcpdm = dev_id; + int irq_status; + + irq_status = omap_mcpdm_read(mcpdm, MCPDM_REG_IRQSTATUS); + + /* Acknowledge irq event */ + omap_mcpdm_write(mcpdm, MCPDM_REG_IRQSTATUS, irq_status); + + if (irq_status & MCPDM_DN_IRQ_FULL) + dev_dbg(mcpdm->dev, "DN (playback) FIFO Full\n"); + + if (irq_status & MCPDM_DN_IRQ_EMPTY) + dev_dbg(mcpdm->dev, "DN (playback) FIFO Empty\n"); + + if (irq_status & MCPDM_DN_IRQ) + dev_dbg(mcpdm->dev, "DN (playback) write request\n"); + + if (irq_status & MCPDM_UP_IRQ_FULL) + dev_dbg(mcpdm->dev, "UP (capture) FIFO Full\n"); + + if (irq_status & MCPDM_UP_IRQ_EMPTY) + dev_dbg(mcpdm->dev, "UP (capture) FIFO Empty\n"); + + if (irq_status & MCPDM_UP_IRQ) + dev_dbg(mcpdm->dev, "UP (capture) write request\n"); + + return IRQ_HANDLED; +} + +static int omap_mcpdm_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); + + mutex_lock(&mcpdm->mutex); + + if (!dai->active) { + pm_runtime_get_sync(mcpdm->dev); + + /* Enable watch dog for ES above ES 1.0 to avoid saturation */ + if (omap_rev() != OMAP4430_REV_ES1_0) { + u32 ctrl = omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL); + + omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, + ctrl | MCPDM_WD_EN); + } + omap_mcpdm_open_streams(mcpdm); + } + + mutex_unlock(&mcpdm->mutex); + + return 0; +} + +static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); + + mutex_lock(&mcpdm->mutex); + + if (!dai->active) { + if (omap_mcpdm_active(mcpdm)) { + omap_mcpdm_stop(mcpdm); + omap_mcpdm_close_streams(mcpdm); + } + + if (!omap_mcpdm_active(mcpdm)) + pm_runtime_put_sync(mcpdm->dev); + } + + mutex_unlock(&mcpdm->mutex); +} + +static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); + int stream = substream->stream; + struct omap_pcm_dma_data *dma_data; + int channels; + int link_mask = 0; + + channels = params_channels(params); + switch (channels) { + case 4: + if (stream == SNDRV_PCM_STREAM_CAPTURE) + /* up to 2 channels for capture */ + return -EINVAL; + link_mask |= 1 << 3; + case 3: + if (stream == SNDRV_PCM_STREAM_CAPTURE) + /* up to 2 channels for capture */ + return -EINVAL; + link_mask |= 1 << 2; + case 2: + link_mask |= 1 << 1; + case 1: + link_mask |= 1 << 0; + break; + default: + /* unsupported number of channels */ + return -EINVAL; + } + + dma_data = &omap_mcpdm_dai_dma_params[stream]; + + /* Configure McPDM channels, and DMA packet size */ + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + mcpdm->dn_channels = link_mask << 3; + dma_data->packet_size = + (MCPDM_DN_THRES_MAX - mcpdm->dn_threshold) * channels; + } else { + mcpdm->up_channels = link_mask << 0; + dma_data->packet_size = mcpdm->up_threshold * channels; + } + + snd_soc_dai_set_dma_data(dai, substream, dma_data); + + return 0; +} + +static int omap_mcpdm_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); + + if (!omap_mcpdm_active(mcpdm)) { + omap_mcpdm_start(mcpdm); + omap_mcpdm_reg_dump(mcpdm); + } + + return 0; +} + +static struct snd_soc_dai_ops omap_mcpdm_dai_ops = { + .startup = omap_mcpdm_dai_startup, + .shutdown = omap_mcpdm_dai_shutdown, + .hw_params = omap_mcpdm_dai_hw_params, + .prepare = omap_mcpdm_prepare, +}; + +static int omap_mcpdm_probe(struct snd_soc_dai *dai) +{ + struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); + int ret; + + pm_runtime_enable(mcpdm->dev); + + /* Disable lines while request is ongoing */ + pm_runtime_get_sync(mcpdm->dev); + omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, 0x00); + + ret = request_irq(mcpdm->irq, omap_mcpdm_irq_handler, + 0, "McPDM", (void *)mcpdm); + + pm_runtime_put_sync(mcpdm->dev); + + if (ret) { + dev_err(mcpdm->dev, "Request for IRQ failed\n"); + pm_runtime_disable(mcpdm->dev); + } + + /* Configure McPDM threshold values */ + mcpdm->dn_threshold = 2; + mcpdm->up_threshold = MCPDM_UP_THRES_MAX - 3; + return ret; +} + +static int omap_mcpdm_remove(struct snd_soc_dai *dai) +{ + struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); + + free_irq(mcpdm->irq, (void *)mcpdm); + pm_runtime_disable(mcpdm->dev); + + return 0; +} + +#define OMAP_MCPDM_RATES (SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) +#define OMAP_MCPDM_FORMATS SNDRV_PCM_FMTBIT_S32_LE + +static struct snd_soc_dai_driver omap_mcpdm_dai = { + .probe = omap_mcpdm_probe, + .remove = omap_mcpdm_remove, + .probe_order = SND_SOC_COMP_ORDER_LATE, + .remove_order = SND_SOC_COMP_ORDER_EARLY, + .playback = { + .channels_min = 1, + .channels_max = 4, + .rates = OMAP_MCPDM_RATES, + .formats = OMAP_MCPDM_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = OMAP_MCPDM_RATES, + .formats = OMAP_MCPDM_FORMATS, + }, + .ops = &omap_mcpdm_dai_ops, +}; + +static __devinit int asoc_mcpdm_probe(struct platform_device *pdev) +{ + struct omap_mcpdm *mcpdm; + struct resource *res; + int ret = 0; + + mcpdm = kzalloc(sizeof(struct omap_mcpdm), GFP_KERNEL); + if (!mcpdm) + return -ENOMEM; + + platform_set_drvdata(pdev, mcpdm); + + mutex_init(&mcpdm->mutex); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (res == NULL) { + dev_err(&pdev->dev, "no resource\n"); + goto err_res; + } + + if (!request_mem_region(res->start, resource_size(res), "McPDM")) { + ret = -EBUSY; + goto err_res; + } + + mcpdm->io_base = ioremap(res->start, resource_size(res)); + if (!mcpdm->io_base) { + ret = -ENOMEM; + goto err_iomap; + } + + mcpdm->irq = platform_get_irq(pdev, 0); + if (mcpdm->irq < 0) { + ret = mcpdm->irq; + goto err_irq; + } + + mcpdm->dev = &pdev->dev; + + ret = snd_soc_register_dai(&pdev->dev, &omap_mcpdm_dai); + if (!ret) + return 0; + +err_irq: + iounmap(mcpdm->io_base); +err_iomap: + release_mem_region(res->start, resource_size(res)); +err_res: + kfree(mcpdm); + return ret; +} + +static int __devexit asoc_mcpdm_remove(struct platform_device *pdev) +{ + struct omap_mcpdm *mcpdm = platform_get_drvdata(pdev); + struct resource *res; + + snd_soc_unregister_dai(&pdev->dev); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + iounmap(mcpdm->io_base); + release_mem_region(res->start, resource_size(res)); + + kfree(mcpdm); + return 0; +} + +static struct platform_driver asoc_mcpdm_driver = { + .driver = { + .name = "omap-mcpdm", + .owner = THIS_MODULE, + }, + + .probe = asoc_mcpdm_probe, + .remove = __devexit_p(asoc_mcpdm_remove), +}; + +static int __init snd_omap_mcpdm_init(void) +{ + return platform_driver_register(&asoc_mcpdm_driver); +} +module_init(snd_omap_mcpdm_init); + +static void __exit snd_omap_mcpdm_exit(void) +{ + platform_driver_unregister(&asoc_mcpdm_driver); +} +module_exit(snd_omap_mcpdm_exit); + +MODULE_AUTHOR("Misael Lopez Cruz <misael.lopez(a)ti.com>"); +MODULE_DESCRIPTION("OMAP PDM SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-mcpdm.h b/sound/soc/omap/omap-mcpdm.h new file mode 100644 index 0000000..d23122a --- /dev/null +++ b/sound/soc/omap/omap-mcpdm.h @@ -0,0 +1,95 @@ +/* + * omap-mcpdm.h + * + * Copyright (C) 2009 - 2011 Texas Instruments + * + * Contact: Misael Lopez Cruz <misael.lopez(a)ti.com> + * + * 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 __OMAP_MCPDM_H__ +#define __OMAP_MCPDM_H__ + +#define MCPDM_REG_REVISION 0x00 +#define MCPDM_REG_SYSCONFIG 0x10 +#define MCPDM_REG_IRQSTATUS_RAW 0x24 +#define MCPDM_REG_IRQSTATUS 0x28 +#define MCPDM_REG_IRQENABLE_SET 0x2C +#define MCPDM_REG_IRQENABLE_CLR 0x30 +#define MCPDM_REG_IRQWAKE_EN 0x34 +#define MCPDM_REG_DMAENABLE_SET 0x38 +#define MCPDM_REG_DMAENABLE_CLR 0x3C +#define MCPDM_REG_DMAWAKEEN 0x40 +#define MCPDM_REG_CTRL 0x44 +#define MCPDM_REG_DN_DATA 0x48 +#define MCPDM_REG_UP_DATA 0x4C +#define MCPDM_REG_FIFO_CTRL_DN 0x50 +#define MCPDM_REG_FIFO_CTRL_UP 0x54 +#define MCPDM_REG_DN_OFFSET 0x58 + +/* + * MCPDM_IRQ bit fields + * IRQSTATUS_RAW, IRQSTATUS, IRQENABLE_SET, IRQENABLE_CLR + */ + +#define MCPDM_DN_IRQ (1 << 0) +#define MCPDM_DN_IRQ_EMPTY (1 << 1) +#define MCPDM_DN_IRQ_ALMST_EMPTY (1 << 2) +#define MCPDM_DN_IRQ_FULL (1 << 3) + +#define MCPDM_UP_IRQ (1 << 8) +#define MCPDM_UP_IRQ_EMPTY (1 << 9) +#define MCPDM_UP_IRQ_ALMST_FULL (1 << 10) +#define MCPDM_UP_IRQ_FULL (1 << 11) + +#define MCPDM_DOWNLINK_IRQ_MASK 0x00F +#define MCPDM_UPLINK_IRQ_MASK 0xF00 + +/* + * MCPDM_DMAENABLE bit fields + */ + +#define MCPDM_DMA_DN_ENABLE (1 << 0) +#define MCPDM_DMA_UP_ENABLE (1 << 1) + +/* + * MCPDM_CTRL bit fields + */ + +#define MCPDM_PDM_UPLINK_EN(x) (1 << (x - 1)) /* ch1 is at bit 0 */ +#define MCPDM_PDM_DOWNLINK_EN(x) (1 << (x + 2)) /* ch1 is at bit 3 */ +#define MCPDM_PDMOUTFORMAT (1 << 8) +#define MCPDM_CMD_INT (1 << 9) +#define MCPDM_STATUS_INT (1 << 10) +#define MCPDM_SW_UP_RST (1 << 11) +#define MCPDM_SW_DN_RST (1 << 12) +#define MCPDM_WD_EN (1 << 14) +#define MCPDM_PDM_UP_MASK 0x7 +#define MCPDM_PDM_DN_MASK (0x1f << 3) + + +#define MCPDM_PDMOUTFORMAT_LJUST (0 << 8) +#define MCPDM_PDMOUTFORMAT_RJUST (1 << 8) + +/* + * MCPDM_FIFO_CTRL bit fields + */ + +#define MCPDM_UP_THRES_MAX 0xF +#define MCPDM_DN_THRES_MAX 0xF + +#endif /* End of __OMAP_MCPDM_H__ */ diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index eec77fe..d462f1a 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -32,7 +32,7 @@ #include <plat/hardware.h> #include <plat/mux.h> -#include "mcpdm.h" +#include "omap-mcpdm.h" #include "omap-pcm.h" #include "../codecs/twl6040.h" -- 1.7.6.1

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[alsa-devel] ALSA: hda: hdmi: Hint matching between input devices and pcm devices
by David Henningsson 30 Aug '11

30 Aug '11

Since modern HDMI cards often have more than one output pin and thus input device, we need to know which one has actually been plugged in. This patch adds a name hint that indicates which PCM device is connected to which pin. To do that, the jack creation has been deferred to build_controls, i e, after the PCM devices have been created. Would be great to have Stephen look through this patch quickly before it's committed. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

4 10

[alsa-devel] Invitation to connect on LinkedIn
by Kuldeep Ghan 30 Aug '11

30 Aug '11

I'd like to add you to my professional network on LinkedIn. - Kuldeep Kuldeep Ghan Software Engineer at Motorola Mobility, Inc Pune Area, India Confirm that you know Kuldeep Ghan: https://www.linkedin.com/e/jpxkk3-gryhzwxz-4y/isd/4021940516/s47LuaNu/ -- You are receiving Invitation to Connect emails. Click to unsubscribe: http://www.linkedin.com/e/jpxkk3-gryhzwxz-4y/uii-rtptP1H4ViCg9YiBUOvto1r4_5… (c) 2011 LinkedIn Corporation. 2029 Stierlin Ct, Mountain View, CA 94043, USA.

1 0

[alsa-devel] [PATCH 3/3] arm: mxs: disable clock-gates when setting saif-clocks
by Dong Aisheng 29 Aug '11

29 Aug '11

From: Dong Aisheng-B29396 <B29396(a)freescale.com> New divides should only be written when gates are off. Reported-by: Dong Aisheng <b29396(a)freescale.com> Signed-off-by: Wolfram Sang <w.sang(a)pengutronix.de> --- BTW, i did a minus change based on wolfram's patch or the saif will not work. Change + __raw_writel(clkgate, CLKCTRL_BASE_ADDR + HW_CLKCTRL_##rs##_SET); \ to + __raw_writel(reg & ~clkgate, CLKCTRL_BASE_ADDR + HW_CLKCTRL_##rs##_SET); \ It seemed HW_CLKCTRL_##rs##_SET did not work well. (i did not find HW_CLKCTRL_SAIFx_SET in spec). --- arch/arm/mach-mxs/clock-mx28.c | 7 +++++-- arch/arm/mach-mxs/regs-clkctrl-mx28.h | 2 ++ 2 files changed, 7 insertions(+), 2 deletions(-) diff --git a/arch/arm/mach-mxs/clock-mx28.c b/arch/arm/mach-mxs/clock-mx28.c index 2a2db65..d1d119a 100644 --- a/arch/arm/mach-mxs/clock-mx28.c +++ b/arch/arm/mach-mxs/clock-mx28.c @@ -438,7 +438,7 @@ _CLK_SET_RATE1(xbus_clk, XBUS) static int name##_set_rate(struct clk *clk, unsigned long rate) \ { \ u16 div; \ - u32 reg; \ + u32 reg, clkgate; \ u64 lrate; \ unsigned long parent_rate; \ int i; \ @@ -455,7 +455,8 @@ static int name##_set_rate(struct clk *clk, unsigned long rate) \ return -EINVAL; \ \ reg = __raw_readl(CLKCTRL_BASE_ADDR + HW_CLKCTRL_##rs); \ - reg &= ~BM_CLKCTRL_##rs##_DIV; \ + clkgate = reg & BM_CLKCTRL_##rs##_CLKGATE; \ + reg &= ~(BM_CLKCTRL_##rs##_DIV | BM_CLKCTRL_##rs##_CLKGATE); \ reg |= div << BP_CLKCTRL_##rs##_DIV; \ __raw_writel(reg, CLKCTRL_BASE_ADDR + HW_CLKCTRL_##rs); \ \ @@ -468,6 +469,8 @@ static int name##_set_rate(struct clk *clk, unsigned long rate) \ return -ETIMEDOUT; \ } \ \ + __raw_writel(reg & ~clkgate, \ + CLKCTRL_BASE_ADDR + HW_CLKCTRL_##rs); \ return 0; \ } diff --git a/arch/arm/mach-mxs/regs-clkctrl-mx28.h b/arch/arm/mach-mxs/regs-clkctrl-mx28.h index 7d1b061..08749b3 100644 --- a/arch/arm/mach-mxs/regs-clkctrl-mx28.h +++ b/arch/arm/mach-mxs/regs-clkctrl-mx28.h @@ -285,6 +285,7 @@ (((v) << 0) & BM_CLKCTRL_EMI_DIV_EMI) #define HW_CLKCTRL_SAIF0 (0x00000100) +#define HW_CLKCTRL_SAIF0_SET (0x00000104) #define BP_CLKCTRL_SAIF0_CLKGATE 31 #define BM_CLKCTRL_SAIF0_CLKGATE 0x80000000 @@ -296,6 +297,7 @@ (((v) << 0) & BM_CLKCTRL_SAIF0_DIV) #define HW_CLKCTRL_SAIF1 (0x00000110) +#define HW_CLKCTRL_SAIF1_SET (0x00000114) #define BP_CLKCTRL_SAIF1_CLKGATE 31 #define BM_CLKCTRL_SAIF1_CLKGATE 0x80000000 -- 1.7.0.4

3 3

Re: [alsa-devel] [PATCH v2] sound: usb: mixer: increase control transfer timeout
by Jassi Brar 29 Aug '11

29 Aug '11

On Mon, Aug 29, 2011 at 6:30 PM, Felipe Balbi <balbi(a)ti.com> wrote: > Hi, > > On Mon, Aug 29, 2011 at 06:07:30PM +0530, Jassi Brar wrote: >> On Mon, Aug 29, 2011 at 6:01 PM, Felipe Balbi <balbi(a)ti.com> wrote: >> > 100ms might not be enough for slower devices. Increasing >> > timeout to 1000ms doesn't seem like overkill. >> > >> > Signed-off-by: Felipe Balbi <balbi(a)ti.com> >> > --- >> > >> > Changes since v1: >> > . Increase all 100ms timeout to 1000ms >> > >> > sound/usb/mixer.c | 4 ++-- >> > sound/usb/mixer_quirks.c | 10 +++++----- >> > 2 files changed, 7 insertions(+), 7 deletions(-) >> > >> > diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c >> > index cdd19d7..1c9d443 100644 >> > --- a/sound/usb/mixer.c >> > +++ b/sound/usb/mixer.c >> > @@ -296,7 +296,7 @@ static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request, int v >> > if (snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), request, >> > USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, >> > validx, snd_usb_ctrl_intf(chip) | (cval->id << 8), >> > - buf, val_len, 100) >= val_len) { >> > + buf, val_len, 1000) >= val_len) { >> > *value_ret = convert_signed_value(cval, snd_usb_combine_bytes(buf, val_len)); >> > snd_usb_autosuspend(cval->mixer->chip); >> > return 0; >> > @@ -445,7 +445,7 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, >> > usb_sndctrlpipe(chip->dev, 0), request, >> > USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, >> > validx, snd_usb_ctrl_intf(chip) | (cval->id << 8), >> > - buf, val_len, 100) >= 0) { >> > + buf, val_len, 1000) >= 0) { >> > snd_usb_autosuspend(chip); >> > return 0; >> > } >> > diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c >> > index 3d0f487..7c41399 100644 >> > --- a/sound/usb/mixer_quirks.c >> > +++ b/sound/usb/mixer_quirks.c >> > @@ -190,18 +190,18 @@ static int snd_audigy2nx_led_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e >> > err = snd_usb_ctl_msg(mixer->chip->dev, >> > usb_sndctrlpipe(mixer->chip->dev, 0), 0x24, >> > USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, >> > - !value, 0, NULL, 0, 100); >> > + !value, 0, NULL, 0, 1000); >> > /* USB X-Fi S51 Pro */ >> > if (mixer->chip->usb_id == USB_ID(0x041e, 0x30df)) >> > err = snd_usb_ctl_msg(mixer->chip->dev, >> > usb_sndctrlpipe(mixer->chip->dev, 0), 0x24, >> > USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, >> > - !value, 0, NULL, 0, 100); >> > + !value, 0, NULL, 0, 1000); >> > else >> > err = snd_usb_ctl_msg(mixer->chip->dev, >> > usb_sndctrlpipe(mixer->chip->dev, 0), 0x24, >> > USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, >> > - value, index + 2, NULL, 0, 100); >> > + value, index + 2, NULL, 0, 1000); >> > if (err < 0) >> > return err; >> > mixer->audigy2nx_leds[index] = value; >> > @@ -299,7 +299,7 @@ static void snd_audigy2nx_proc_read(struct snd_info_entry *entry, >> > usb_rcvctrlpipe(mixer->chip->dev, 0), >> > UAC_GET_MEM, USB_DIR_IN | USB_TYPE_CLASS | >> > USB_RECIP_INTERFACE, 0, >> > - jacks[i].unitid << 8, buf, 3, 100); >> > + jacks[i].unitid << 8, buf, 3, 1000); >> > if (err == 3 && (buf[0] == 3 || buf[0] == 6)) >> > snd_iprintf(buffer, "%02x %02x\n", buf[1], buf[2]); >> > else >> > @@ -332,7 +332,7 @@ static int snd_xonar_u1_switch_put(struct snd_kcontrol *kcontrol, >> > err = snd_usb_ctl_msg(mixer->chip->dev, >> > usb_sndctrlpipe(mixer->chip->dev, 0), 0x08, >> > USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, >> > - 50, 0, &new_status, 1, 100); >> > + 50, 0, &new_status, 1, 1000); >> >> How about also converting to use a local define for timeout value ? > > should it be part of another patch ? The current code has no defines at > all and $SUBJECT is just changing what's currently there to another > value. Can't speak for maintainers, but personally I'd prefer a single patch with two points in changelog. The change is trivial enough.

1 0

Re: [alsa-devel] [PATCH] sound: usb: mixer: increase control transfer timeout
by Clemens Ladisch 29 Aug '11

29 Aug '11

Felipe Balbi wrote: > 100ms might not be enough for slower devices. Increasing > timeout to 1000ms doesn't seem like overkill. > > --- a/sound/usb/mixer.c > +++ b/sound/usb/mixer.c > @@ -296,7 +296,7 @@ static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request, int v > if (snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), request, > USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, > validx, snd_usb_ctrl_intf(chip) | (cval->id << 8), > - buf, val_len, 100) >= val_len) { > + buf, val_len, 1000) >= val_len) { If the timeout for reading values needs to be increased, writing (in snd_usb_mixer_set_ctl_value) probably needs some attention too. Regards, Clemens

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Re: [alsa-devel] Can ALSA invoke a Callback ?
by Clemens Ladisch 29 Aug '11

29 Aug '11

Ru Vuott wrote: > you wrote: > > These functions are designed to be used with poll(); to use > > them with Gambas, extract the file descriptors and their event masks > > from the pfds array and call GB.Watch() with these values. > > << GB.Watch() is an internal interpreter API. You can't use it in Gambas code.>> Then how do you get access to the snd_seq_* functions? I assumed that you have some module that offers a Gambas API on top of the ALSA C API; that module then should also wrap the snd_seq_poll_* calls. > I need a "file", so I could use in Gambas: > > hfile = Open "....." for Read W A T C H You would need some mechanism to watch a file that is identified by its already opened file descriptor, not by its name. It looks as if the only such mechanism is something that uses the internal interpreter API. Regards, Clemens

1 0

Re: [alsa-devel] [PATCH] ASoC: Add 3D stereo support for wm8996
by Mark Brown 29 Aug '11

29 Aug '11

On Fri, Aug 26, 2011 at 12:14:14PM -0700, susan gao wrote: > My first patch to ASoC ever! If I did something wrong, blame Ian. Applied, thanks.

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[alsa-devel] [PATCH] ASoC: davinci-mcasp: add support for unsigned PCM formats
by Ben Gardiner 29 Aug '11

29 Aug '11

Although the McASP supports sign-extending samples in RX or TX [1]; the davinci-mcasp driver does not touch the {R,X}PBIT or {R,X}PAD field of the {R,X}FMT registers meaning that the McASP will serialize the bytes it is given regardless of their signedness. So supporting unsigned formats is as simple as adding them to the metadata of the davinci-mcasp driver. Update the FMTBITs reported in the snd_soc_dai_driver and also update the case statements in davinci-mcasp's hw_params() function so that the McASP can be connected to CODECs that use unsigned values. [1] http://www.ti.com/lit/ug/sprufm1/sprufm1.pdf Signed-off-by: Ben Gardiner <bengardiner(a)nanometrics.ca> --- Tested with a logic analyzer on da850evm where an unsigned 16bit codec was registered with the mcasp and a 16bit signed file was played to it. ALSA's plug pcm mapped to unsigned and the samples serialized as unsigned 16bit -- as expected. --- sound/soc/davinci/davinci-mcasp.c | 20 +++++++++++++------- 1 files changed, 13 insertions(+), 7 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 8566238..7173df2 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -732,16 +732,19 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, davinci_hw_param(dev, substream->stream); switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U8: case SNDRV_PCM_FORMAT_S8: dma_params->data_type = 1; word_length = DAVINCI_AUDIO_WORD_8; break; + case SNDRV_PCM_FORMAT_U16_LE: case SNDRV_PCM_FORMAT_S16_LE: dma_params->data_type = 2; word_length = DAVINCI_AUDIO_WORD_16; break; + case SNDRV_PCM_FORMAT_U32_LE: case SNDRV_PCM_FORMAT_S32_LE: dma_params->data_type = 4; word_length = DAVINCI_AUDIO_WORD_32; @@ -818,6 +821,13 @@ static struct snd_soc_dai_ops davinci_mcasp_dai_ops = { }; +#define DAVINCI_MCASP_PCM_FMTS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_U8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S32_LE | \ + SNDRV_PCM_FMTBIT_U32_LE) + static struct snd_soc_dai_driver davinci_mcasp_dai[] = { { .name = "davinci-mcasp.0", @@ -825,17 +835,13 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { .channels_min = 2, .channels_max = 2, .rates = DAVINCI_MCASP_RATES, - .formats = SNDRV_PCM_FMTBIT_S8 | - SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S32_LE, + .formats = DAVINCI_MCASP_PCM_FMTS, }, .capture = { .channels_min = 2, .channels_max = 2, .rates = DAVINCI_MCASP_RATES, - .formats = SNDRV_PCM_FMTBIT_S8 | - SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S32_LE, + .formats = DAVINCI_MCASP_PCM_FMTS, }, .ops = &davinci_mcasp_dai_ops, @@ -846,7 +852,7 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { .channels_min = 1, .channels_max = 384, .rates = DAVINCI_MCASP_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + .formats = DAVINCI_MCASP_PCM_FMTS, }, .ops = &davinci_mcasp_dai_ops, }, -- 1.7.4.1

3 2

[alsa-devel] support for IDT 92HD89E codec
by Andreas Berglund 28 Aug '11

28 Aug '11

Hi! I have a sounccard with the IDT 92HD89E codec, or at least I thought so, when I run the alsa-info script I get the IDT ID 76c7 returned as the codec. However none of them seem to be supported by alsa, are there plans to support these codecs in the future? I am running the latest version of debian stable and have version 1.0.21 of the driver and version 1.0.23 of the library and utilities running. regards Andreas Berglund

2 2

Re: [alsa-devel] [PATCH] ASoC: Add 3D stereo support for wm8996
by Liam Girdwood 28 Aug '11

28 Aug '11

On 26/08/11 20:14, susan gao wrote: > My first patch to ASoC ever! If I did something wrong, blame Ian. > No blame required ;) > Signed-off-by: Susan Gao <sgao(a)opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg(a)ti.com> > --- > sound/soc/codecs/wm8996.c | 9 +++++++++ > 1 files changed, 9 insertions(+), 0 deletions(-) > > diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c > index 73a1588..c4bc99b 100644 > --- a/sound/soc/codecs/wm8996.c > +++ b/sound/soc/codecs/wm8996.c > @@ -414,6 +414,7 @@ static const DECLARE_TLV_DB_SCALE(out_digital_tlv, -1200, 150, 0); > static const DECLARE_TLV_DB_SCALE(out_tlv, -900, 75, 0); > static const DECLARE_TLV_DB_SCALE(spk_tlv, -900, 150, 0); > static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); > +static const DECLARE_TLV_DB_SCALE(threedstereo_tlv, -1600, 183, 1); > > static const char *sidetone_hpf_text[] = { > "2.9kHz", "1.5kHz", "735Hz", "403Hz", "196Hz", "98Hz", "49Hz" > @@ -608,6 +609,14 @@ SOC_SINGLE("DAC High Performance Switch", WM8996_OVERSAMPLING, 0, 1, 0), > SOC_SINGLE("DAC Soft Mute Switch", WM8996_DAC_SOFTMUTE, 1, 1, 0), > SOC_SINGLE("DAC Slow Soft Mute Switch", WM8996_DAC_SOFTMUTE, 0, 1, 0), > > +SOC_SINGLE("DSP1 3D Stereo Switch", WM8996_DSP1_RX_FILTERS_2, 8, 1, 0), > +SOC_SINGLE("DSP2 3D Stereo Switch", WM8996_DSP2_RX_FILTERS_2, 8, 1, 0), > + > +SOC_SINGLE_TLV("DSP1 3D Stereo Volume", WM8996_DSP1_RX_FILTERS_2, 10, 15, > + 0, threedstereo_tlv), > +SOC_SINGLE_TLV("DSP2 3D Stereo Volume", WM8996_DSP2_RX_FILTERS_2, 10, 15, > + 0, threedstereo_tlv), > + > SOC_DOUBLE_TLV("Digital Output 1 Volume", WM8996_DAC1_HPOUT1_VOLUME, 0, 4, > 8, 0, out_digital_tlv), > SOC_DOUBLE_TLV("Digital Output 2 Volume", WM8996_DAC2_HPOUT2_VOLUME, 0, 4,

1 0

[alsa-devel] How to add a new MIDI device to the usbaudio driver
by Kristian Amlie 27 Aug '11

27 Aug '11

I have just received a new MIDI instrument (Starr Labs Z7S) which is not recognized by the kernel, and I'd like to add support for it. I'm hoping that it uses a standard MIDI interface and it's just a matter of adding the correct entry in the table of devices. I have tried to add an entry to sound/usb/usbquirks.h (sound/usb/quirks-table.h in later kernels), but so far I haven't succeeded. Is there any documentation for how to do this, and which options to use? Google hasn't really turned up anything... -- Kristian

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Re: [alsa-devel] M-Audio FTU remaining mixer controls and external clocking support.
by Daniel Mack 27 Aug '11

27 Aug '11

On Fri, Aug 26, 2011 at 4:06 PM, Grant Diffey <gdiffey(a)gmail.com> wrote: > On Fri, Aug 26, 2011 at 11:43 PM, Daniel Mack <zonque(a)gmail.com> wrote: >> >> Can you try to just open either the playback or the capture stream >> seperately, and not both at the same time? > > So... > > nevyn@cetacea:~/linux/linux$ aplay -D plughw:Ultra > /usr/share/sounds/KDE_Startup_new.wav > Playing WAVE '/usr/share/sounds/KDE_Startup_new.wav' : Signed 16 bit Little > Endian, Rate 22050 Hz, Stereo > aplay: set_params:1123: Unable to install hw params: Hmm? Strange, that worked for me. Does re-plugging the device help maybe? I can look into this deeper next week. The patch I sent out was also merely meant to be a base for discussion about whether the whole approach is sane, or just a result of my head thinking about it for too long :) Daniel

2 1

Re: [alsa-devel] Can ALSA invoke a Callback ?
by Clemens Ladisch 27 Aug '11

27 Aug '11

Ru Vuott wrote: > Well, I 'm programming with Gambas 3.0 language. I 'm using > a GUI. Use snd_seq_poll_descriptors_count() and snd_seq_poll_descriptors() to get one or more file descriptors and event masks for the sequencer device; these file descriptors can be used to wait for events. When you receive an event, check if the device is actually ready for reading or writing by calling snd_seq_poll_descriptors_revents(). These functions are designed to be used with poll(); to use them with Gambas, extract the file descriptors and their event masks from the pfds array and call GB.Watch() with these values. Regards, Clemens

1 0

[alsa-devel] TASCAM US-122mkII
by Cyril Hrubis 26 Aug '11

26 Aug '11

Hi! The good news is that I have 122mkII to play with, the bad news is that it doesn't work with the snd_usb_122l driver. The probe function fails most likely in usb_stream_start(&us122l->sk), The dmesg output: ... status=-2 status=-2 us122l_start error -14 snd-usb-us122l: probe of 1-2:1.1 failed with error -22 usbcore: registered new interface driver snd-usb-us122l ... I've set up a windows machine with usbsnoop and captured the init log (I could upload it somewhere if anybody is interested). I would probably be able to help fix/write the driver with some guidance. Just now I'm trying to understand the log... PS: I've send this mail to alsa-devel a couple of days ago, but without being subscribed, does anybody do the screening on incomming mails? -- Cyril Hrubis

3 9

Re: [alsa-devel] Can ALSA invoke a Callback ?
by Clemens Ladisch 26 Aug '11

26 Aug '11

Ru Vuott wrote: > I'm using Alsa "Seq" functions. > > My midi-program - obviously - does not know exactly when the extern midi event 'll arrive.... > Has it to read uninterruptedly his port, waiting for the events.... ? This is done only in single-purpose command-line programs like arecordmidi. When you have other events, you collect the file descriptors of all your devices/pipes/whatever, and wait for all of them in a poll()-based loop. How does your program handle events? Or are you using a GUI framework? Regards, Clemens

1 0

[alsa-devel] Can ALSA invoke a Callback ?
by Ru Vuott 26 Aug '11

26 Aug '11

Hello, I would like to know how I can ask ALSA to invoke a "Callback" function when an Midi event arrives from an extern-device. Thanks Paolo

2 1

Re: [alsa-devel] M-Audio FTU remaining mixer controls and external clocking support.
by Daniel Mack 26 Aug '11

26 Aug '11

On Aug 26, 2011 3:35 PM, "Grant Diffey" <gdiffey(a)gmail.com> wrote: > > So on my freshly built 3.1-rc1 this applied cleanly but I'm getting errors every time I try to open the device. on the FTU. > > attempting to start jack gives this in dmesg: > > [ 2091.951806] stop_endpoints() > [ 2197.403224] Re-using EP 81 @ffff880003560000 > [ 2197.403243] attr 4 > [ 2197.403929] stop_endpoints() > [ 2197.403931] -- data ffff880003560000 > [ 2197.403933] -- sync (null) > [ 2197.403935] Unable to change format on ep #81: already in use > > and hitting play on audacity results in the attached dmesg errors. > > is this expected currently? Can you try to just open either the playback or the capture stream seperately, and not both at the same time? Daniel

2 1

[alsa-devel] [RFC PATCH 1/1] ASoC: soc-core: symmetry checking for each DAIs separately
by Dong Aisheng 26 Aug '11

26 Aug '11

The orginal code does not cover the case that one DAI such as codec may be shared between other two DAIs(CPU). When do symmetry checking, altough the codec DAI requires symmetry, the two CPU DAIs may still be configured to run on different rates. We change to check each DAI's state separately instead of only checking the dai link to prevent this issue. 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> Cc: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc-dai.h | 3 ++ include/sound/soc.h | 2 - sound/soc/soc-pcm.c | 64 +++++++++++++++++++++++++++++++++------------- 3 files changed, 49 insertions(+), 20 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 5ad5f3a..844d4ed 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -242,6 +242,9 @@ struct snd_soc_dai { void *playback_dma_data; void *capture_dma_data; + /* Symmetry data */ + unsigned int rate; + /* parent platform/codec */ union { struct snd_soc_platform *platform; diff --git a/include/sound/soc.h b/include/sound/soc.h index 3fe658e..5449139 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -849,8 +849,6 @@ struct snd_soc_pcm_runtime { unsigned int complete:1; unsigned int dev_registered:1; - /* Symmetry data - only valid if symmetry is being enforced */ - unsigned int rate; long pmdown_time; /* runtime devices */ diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 1aee9fc..3f7ded7 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -32,33 +32,54 @@ static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai = rtd->codec_dai; + unsigned int race; + unsigned int force_rate; int ret; + race = 0; + force_rate = 0; + if (!codec_dai->driver->symmetric_rates && !cpu_dai->driver->symmetric_rates && !rtd->dai_link->symmetric_rates) return 0; + if (codec_dai->active && codec_dai->driver->symmetric_rates || + codec_dai->active && rtd->dai_link->symmetric_rates) { + if (codec_dai->rate != 0) + force_rate = codec_dai->rate; + else + race = 1; + } + + if (cpu_dai->active && cpu_dai->driver->symmetric_rates || + codec_dai->active && rtd->dai_link->symmetric_rates) { + if (cpu_dai->rate != 0) + force_rate = cpu_dai->rate; + else + race = 1; + } + + if (force_rate) { + dev_dbg(&rtd->dev, "Symmetry forces %dHz rate\n", rtd->rate); + + ret = snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + force_rate, force_rate); + if (ret < 0) { + dev_err(&rtd->dev, + "Unable to apply rate symmetry constraint: %d\n", ret); + return ret; + } + } + /* This can happen if multiple streams are starting simultaneously - * the second can need to get its constraints before the first has * picked a rate. Complain and allow the application to carry on. */ - if (!rtd->rate) { + if (race) dev_warn(&rtd->dev, - "Not enforcing symmetric_rates due to race\n"); - return 0; - } - - dev_dbg(&rtd->dev, "Symmetry forces %dHz rate\n", rtd->rate); - - ret = snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_RATE, - rtd->rate, rtd->rate); - if (ret < 0) { - dev_err(&rtd->dev, - "Unable to apply rate symmetry constraint: %d\n", ret); - return ret; - } + "Not enforcing symmetric_rates due to race\n"); return 0; } @@ -287,9 +308,14 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) cpu_dai->active--; codec_dai->active--; codec->active--; + rtd->active--; + + /* clear the corresponding DAIs rate when inactive */ + if (!cpu_dai->active) + cpu_dai->rate = 0; - if (!cpu_dai->active && !codec_dai->active) - rtd->rate = 0; + if (!codec_dai->active) + codec_dai->rate = 0; /* Muting the DAC suppresses artifacts caused during digital * shutdown, for example from stopping clocks. @@ -447,7 +473,9 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } } - rtd->rate = params_rate(params); + /* store the rate for each DAIs */ + cpu_dai->rate = params_rate(params); + codec_dai->rate = params_rate(params); out: mutex_unlock(&rtd->pcm_mutex); -- 1.7.0.4

3 6

[alsa-devel] Will be on vacation
by Takashi Iwai 26 Aug '11

26 Aug '11

Hi, I'll be on vacation from tomorrow for two weeks. As I'll be in a round trip in Sri Lanka, likely unreachable in most time, and I'm going to catch up things after back from vacation. Or, if anyone wants to take a maintenance of 3.1/3.2 sound git trees during my absence, feel free to do that. thanks, Takashi

1 0

[alsa-devel] [PATCH v2 1/1] ASoC: soc-core: check rate for symmetry only when pcm is active
by Dong Aisheng 26 Aug '11

26 Aug '11

For the playback and record using different dai links, checking !rtd->rate for symmetry may not be accurate because that pcm may be acutually not running and the default new open rate is 0, then the warning message "Not enforcing symmetric_rates" will happen each time with running arecord | aplay. Now we only check rate for symmetry when the pcm is really active which seems more sensible. 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> --- include/sound/soc.h | 1 + sound/soc/soc-pcm.c | 5 +++++ 2 files changed, 6 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 3fe658e..17970fd 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -851,6 +851,7 @@ struct snd_soc_pcm_runtime { /* Symmetry data - only valid if symmetry is being enforced */ unsigned int rate; + unsigned int active; long pmdown_time; /* runtime devices */ diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 1aee9fc..72a7b34 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -39,6 +39,9 @@ static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream) !rtd->dai_link->symmetric_rates) return 0; + if (!rtd->active) + return 0; + /* This can happen if multiple streams are starting simultaneously - * the second can need to get its constraints before the first has * picked a rate. Complain and allow the application to carry on. @@ -209,6 +212,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) cpu_dai->active++; codec_dai->active++; rtd->codec->active++; + rtd->active++; mutex_unlock(&rtd->pcm_mutex); return 0; @@ -287,6 +291,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) cpu_dai->active--; codec_dai->active--; codec->active--; + rtd->active--; if (!cpu_dai->active && !codec_dai->active) rtd->rate = 0; -- 1.7.0.4

5 6

[alsa-devel] [PATCH v2 0/2] ASoC: omap-mcpdm: New McPDM driver
by Peter Ujfalusi 26 Aug '11

26 Aug '11

Hello Mark, Liam, I tried to address the comments regarding to version 1 of the new McPDM driver: http://mailman.alsa-project.org/pipermail/alsa-devel/2011-July/041690.html The omap hwmod conversion part of the v1 series has been sent as part of the fixup series for 3.1 to make basic audio work out of box. This series is aimed for 3.2 due to the scale of the change. I tried to split patch 2, but I did not managed to find an incremental path from the old driver to the new one. I have updated the commit message for patch 2 to explain the reasoning behind the rewritten driver, so it can be easily looked up. Notable changes since v1: - delayed_work from the new driver has been removed, and replaced with a DAPM_SUPPLY widget. This widget need to be connected to DAC/ADC of the codec by the machine driver in order to have the correct sequence for power off. The first patch adds private_data pointer for dapm widgets. This private_data can be used to add widget specific data to any widget. I think this can be handy for others as well in the future. - Full duplex audio in legacy mode has been fixed, so we can run arecord | aplay This series depends on the McPDM fixup patches: http://mailman.alsa-project.org/pipermail/alsa-devel/2011-August/042401.html Peter PS: to make the review easier, I have asked git format-patch to treat the change as rewrite. --- Misael Lopez Cruz (1): ASoC: omap-mcpdm: Replace legacy driver Peter Ujfalusi (1): ASoC: DAPM: Add private data pointer for DAPM widget include/sound/soc-dapm.h | 1 + sound/soc/omap/Makefile | 2 +- sound/soc/omap/mcpdm.c | 472 ------------------------- sound/soc/omap/mcpdm.h | 152 -------- sound/soc/omap/omap-mcpdm.c | 823 ++++++++++++++++++++++++++++--------------- sound/soc/omap/omap-mcpdm.h | 97 +++++ sound/soc/omap/sdp4430.c | 14 +- 7 files changed, 656 insertions(+), 905 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

5 21

[alsa-devel] ALSA calls don't work from a shared library
by Olivier Guillion - Myriad 25 Aug '11

25 Aug '11

Hi, I opened a thread yesterday called "ALSA calls don't work within Mozilla plug- ins", but after more testing, I realize that it is a more global problem, so I submit another request for help with a proper title. It seems that several ALSA functions either crash or return unpredictable values when called from a dynamic library (.so). I created a very short program that opens a pcm device, sets the sample rate to 44100 and reads the sample rate back to ensure it's correct : --------------- #include "alsa/asoundlib.h" static char displayString[1024]; char * ALSATest(void) { int err; snd_pcm_t * handle; snd_pcm_hw_params_t *params; unsigned int val,val2; char status[1024]; err = snd_pcm_open(&handle, "plughw:0,0",SND_PCM_STREAM_PLAYBACK, 0); if(err>=0) { snd_pcm_hw_params_alloca(&params); snd_pcm_hw_params_any(handle, params); val=44100; err=snd_pcm_hw_params_set_rate_near(handle,params, &val, NULL); val2=0; err|=snd_pcm_hw_params_get_rate(params, &val2, NULL); if(val!=44100 || val2!=44100 || err!=0) sprintf(status,"ALSA is not working (val2=%d err=%d)",val2,err); else strcpy(status,"ALSA is OK"); snd_pcm_close(handle); } else strcpy(status,"device cannot be opened"); sprintf(displayString,"ALSA %s\nStatus:%s",snd_asoundlib_version(),status); return(displayString); } ------------- If I call this function directly from the main program, it works well (I get "ALSA is OK"). Then if I compile it as a shared library (mysharedlib.so), and in another project I write: --------------- #include <dlfcn.h> #include "alsa/asoundlib.h" typedef char * (*ALSAProc)(void); int main() { void *lib_handle; void *fn; char * str; lib_handle=dlopen("mysharedlib.so",RTLD_NOW); if (lib_handle) { fn=dlsym(lib_handle,"ALSATest"); if(fn) str=((ALSAProc)fn)(); else str="function not found"; } else str="lib not found"; fprintf(stderr,"%s\n",str); if(lib_handle) dlclose(lib_handle); return 0; } ------------- I get unpredictable results. Either; - it crashes in snd_pcm_hw_params_any (in the subroutine snd_pcm_hw_params_refine) or - it outputs "ALSA is not working (val2=0 err=2147483647)" or - it outputs ALSA is not working (val2=2147483647 err=2147483647)" Does anyone have an idea of what could cause this behaviour? Olivier

2 3

Re: [alsa-devel] [PATCH] ALSA: usb: refine delay information with USB frame counter
by Clemens Ladisch 25 Aug '11

25 Aug '11

Pierre-Louis Bossart wrote: > > > + if (frame_diff < 0) > > > + frame_diff += 1024; /* handle 10-bit wrap-around */ > > > > After reading through some HCD source files, I deduced the following > > wrap-arounds: > > > > EHCI: 8-10 bits (default 9) > > FHCI: 11 bits > > IMX21: 16 bits > > ISP1760: 10 bits > > OHCI: 16 bits > > OXU210HP: 10 bits > > R8A66597: 10 bits > > SL811: 5 bits(!?) > > UHCI: 32 bits > > xHCI: ? > > > > As long as there isn't an API that tells us about the valid range of the > > frame counter, we should probably mask off all except the lowest few bits. > > If we only used the 8 least-significant bits, it'd still leave us with up to > 256 ms. That should work I guess. But the SL811 with 5 bits is a problem, > that makes 32ms, this is probably too low. Sorry, I read the code wrong; that driver returns a 16-bit frame counter, it just doesn't allow to schedule more than 32 frame AFAICT. And that driver doesn't support USB audio anyway, so 8 bits should be fine. Regards, Clemens

1 0

[alsa-devel] [PATCH] ALSA: usb: refine delay information with USB frame counter
by Pierre-Louis Bossart 25 Aug '11

25 Aug '11

Existing code only updates the audio delay when URBs were submitted/retired. This can introduce an uncertainty of 8ms on the number of samples played out with the default settings, and a lot more when URBs convey more packets to reduce the interrupt rate and power consumption. This patch relies on the USB frame counter to reduce the uncertainty to ~1ms. The delay information essentially becomes independent of the URB size and number of packets. This should improve audio/video sync and help applications like PulseAudio which require accurate audio timing. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/usb/card.h | 2 ++ sound/usb/pcm.c | 23 +++++++++++++++++++++++ sound/usb/pcm.h | 3 +++ sound/usb/urb.c | 20 +++++++++++++++++--- 4 files changed, 45 insertions(+), 3 deletions(-) diff --git a/sound/usb/card.h b/sound/usb/card.h index ae4251d..a39edcc 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -94,6 +94,8 @@ struct snd_usb_substream { spinlock_t lock; struct snd_urb_ops ops; /* callbacks (must be filled at init) */ + int last_frame_number; /* stored frame number */ + int last_delay; /* stored delay */ }; struct snd_usb_stream { diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index b8dcbf4..1696e2b 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -34,6 +34,27 @@ #include "clock.h" #include "power.h" +/* return the estimated delay based on USB frame counters */ +snd_pcm_uframes_t snd_usb_pcm_delay(struct snd_usb_substream *subs, + unsigned int rate) +{ + int current_frame_number; + int frame_diff; + int est_delay; + + current_frame_number = usb_get_current_frame_number(subs->dev); + frame_diff = current_frame_number-subs->last_frame_number; + if (frame_diff < 0) + frame_diff += 1024; /* handle 10-bit wrap-around */ + + /* Approximation based on number of samples per USB frame (ms), + some truncation for 44.1 but the estimate is good enough */ + est_delay = subs->last_delay-(frame_diff*rate/1000L); + if (est_delay < 0) + est_delay = 0; + return est_delay; +} + /* * return the current pcm pointer. just based on the hwptr_done value. */ @@ -45,6 +66,8 @@ static snd_pcm_uframes_t snd_usb_pcm_pointer(struct snd_pcm_substream *substream subs = (struct snd_usb_substream *)substream->runtime->private_data; spin_lock(&subs->lock); hwptr_done = subs->hwptr_done; + substream->runtime->delay = snd_usb_pcm_delay(subs, + substream->runtime->rate); spin_unlock(&subs->lock); return hwptr_done / (substream->runtime->frame_bits >> 3); } diff --git a/sound/usb/pcm.h b/sound/usb/pcm.h index ed3e283..df7a003 100644 --- a/sound/usb/pcm.h +++ b/sound/usb/pcm.h @@ -1,6 +1,9 @@ #ifndef __USBAUDIO_PCM_H #define __USBAUDIO_PCM_H +snd_pcm_uframes_t snd_usb_pcm_delay(struct snd_usb_substream *subs, + unsigned int rate); + void snd_usb_set_pcm_ops(struct snd_pcm *pcm, int stream); int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface, diff --git a/sound/usb/urb.c b/sound/usb/urb.c index e184349..ccf77d1 100644 --- a/sound/usb/urb.c +++ b/sound/usb/urb.c @@ -718,7 +718,15 @@ static int prepare_playback_urb(struct snd_usb_substream *subs, subs->hwptr_done += bytes; if (subs->hwptr_done >= runtime->buffer_size * stride) subs->hwptr_done -= runtime->buffer_size * stride; + + /* update delay with exact number of samples queued */ + runtime->delay = subs->last_delay; runtime->delay += frames; + subs->last_delay = runtime->delay; + + /* realign last_frame_number */ + subs->last_frame_number = usb_get_current_frame_number(subs->dev); + spin_unlock_irqrestore(&subs->lock, flags); urb->transfer_buffer_length = bytes; if (period_elapsed) @@ -737,12 +745,18 @@ static int retire_playback_urb(struct snd_usb_substream *subs, unsigned long flags; int stride = runtime->frame_bits >> 3; int processed = urb->transfer_buffer_length / stride; + int est_delay; spin_lock_irqsave(&subs->lock, flags); - if (processed > runtime->delay) - runtime->delay = 0; + + est_delay = snd_usb_pcm_delay(subs, runtime->rate); + /* update delay with exact number of samples played */ + if (processed > subs->last_delay) + subs->last_delay = 0; else - runtime->delay -= processed; + subs->last_delay -= processed; + runtime->delay = subs->last_delay; + spin_unlock_irqrestore(&subs->lock, flags); return 0; } -- 1.7.6

2 2

[alsa-devel] M-Audio FTU remaining mixer controls and external clocking support.
by Aurélien Leblond 25 Aug '11

25 Aug '11

>Yes, sorry for this taking longer than expected. I've been working on >this for a while and will continue doing so next week. I just got too >much things to do in parallel these days. I'll get back to you once I >can show patches for testing. That's a great news, I'm ready any time for testing! Aurélien

5 5

[alsa-devel] [PATCH] ALSA: hda: Conexant: Allow different output types to share DAC
by David Henningsson 25 Aug '11

25 Aug '11

[PATCH] ALSA: hda: Conexant: Allow different output types to share DAC Headphones has stopped working for the original reported (a regression compared to 2.6.38). This is because Speaker and Headphones share the same DAC, in which case no Headphones volume control was created. This patch fixes so that both Speaker and Headphones volume controls are created in such scenario. Buglink: http://bugs.launchpad.net/bugs/817943 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> Link to codec-proc: https://launchpadlibrarian.net/76215662/Card0.Codecs.codec.0.txt -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 4

[alsa-devel] ASoC updates for 3.2
by Mark Brown 25 Aug '11

25 Aug '11

The following changes since commit e3d73c1bbf08b9abd3f56293796ba7b5c15008f5: ASoC: sta32x: Move resource allocation and release to the corresponding callback functions (2011-08-19 01:25:56 +0900) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.2 Axel Lin (1): ASoC: soc-core: use GFP_KERNEL flag for kmalloc in snd_soc_cnew Dong Aisheng (2): ASoC: mxs-saif: clear clk gate first before register setting ASoC: mxs-sgtl5000: add record function Joseph Pentland (1): ASoC: Add Springbank I/O card to Speyside Kconfig Julia Lawall (5): sound/soc/kirkwood/kirkwood-i2s.c: add missing kfree sound/soc/ep93xx/ep93xx-i2s.c: add missing kfree sound/soc/fsl/p1022_ds.c: add missing of_node_put sound/soc/fsl/mpc8610_hpcd.c: add missing of_node_put sound/soc/mxs/mxs-saif.c: add missing kfree Mark Brown (18): ASoC: WM8996 record paths need AIFCLK ASoC: Fix configuration of WM8996 input enables ASoC: Automatically manage WM8996 MICBIAS regulating mode ASoC: Optimise WM8996 no interrupt path ASoC: Add basic WM8962 capture low/high pass filter control ASoC: Ensure we only run Speyside WM8962 bias level callbacks once ASoC: Initial WM8962 DSP2 support ASoC: Report IRQ_NONE when we don't see an interrupt from WM8962 ASoC: Clear any outstanding WM8962 FLL lock completions before waiting ASoC: Provide more detail on WM8962 thermal shutdown status ASoC: Clear completions from late WM8996 FLL lock IRQs Merge branch 'for-3.1' into for-3.2 ASoC: Convert WM8523 to table based control and DAPM initialization ASoC: Allow idle_bias_off to be specified in CODEC drivers Merge branch 'for-3.1' into for-3.2 Merge branch 'for-3.1' into for-3.2 ASoC: Correct element count for WM8996 sidetone HPF Merge branch 'for-3.1' into for-3.2 Sangbeom Kim (2): ASoC: Add samsung maintainer ASoC: SAMSUNG: Add Kconfig to support SMDK4212 Stephen Warren (1): ASoC: Tegra: wm8903 machine driver: Drop Ventana support Timur Tabi (2): sound/soc/fsl/fsl_dma.c: add missing of_node_put ASoC: MPC5200: replace of_device with platform_device MAINTAINERS | 1 + include/sound/soc.h | 1 + sound/soc/codecs/wm8523.c | 26 ++--- sound/soc/codecs/wm8962.c | 191 ++++++++++++++++++++++++++++++++++- sound/soc/codecs/wm8996.c | 40 +++++--- sound/soc/ep93xx/ep93xx-i2s.c | 5 +- sound/soc/fsl/fsl_dma.c | 2 + sound/soc/fsl/mpc5200_dma.c | 6 +- sound/soc/fsl/mpc8610_hpcd.c | 18 ++-- sound/soc/fsl/p1022_ds.c | 4 +- sound/soc/kirkwood/kirkwood-i2s.c | 2 +- sound/soc/mxs/mxs-saif.c | 27 +++-- sound/soc/mxs/mxs-sgtl5000.c | 10 ++- sound/soc/samsung/Kconfig | 7 +- sound/soc/samsung/speyside_wm8962.c | 6 + sound/soc/soc-core.c | 4 +- sound/soc/tegra/tegra_wm8903.c | 4 +- 17 files changed, 285 insertions(+), 69 deletions(-)

2 1

[alsa-devel] ASoC updates for 3.1
by Mark Brown 25 Aug '11

25 Aug '11

The following changes since commit e574044acbad7421879270a80acd337459c94cc8: ASoC: omap: Fix build errors in ams-delta (2011-08-19 01:19:45 +0900) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.1 A fairly large collection of small, local patches that mostly only affect individual drivers - should be safe. Axel Lin (1): ASoC: soc-core: use GFP_KERNEL flag for kmalloc in snd_soc_cnew Joseph Pentland (1): ASoC: Add Springbank I/O card to Speyside Kconfig Julia Lawall (4): sound/soc/kirkwood/kirkwood-i2s.c: add missing kfree sound/soc/ep93xx/ep93xx-i2s.c: add missing kfree sound/soc/fsl/p1022_ds.c: add missing of_node_put sound/soc/fsl/mpc8610_hpcd.c: add missing of_node_put Mark Brown (6): ASoC: WM8996 record paths need AIFCLK ASoC: Fix configuration of WM8996 input enables ASoC: Ensure we only run Speyside WM8962 bias level callbacks once ASoC: Clear any outstanding WM8962 FLL lock completions before waiting ASoC: Clear completions from late WM8996 FLL lock IRQs ASoC: Correct element count for WM8996 sidetone HPF Sangbeom Kim (1): ASoC: Add samsung maintainer Stephen Warren (1): ASoC: Tegra: wm8903 machine driver: Drop Ventana support Timur Tabi (1): sound/soc/fsl/fsl_dma.c: add missing of_node_put MAINTAINERS | 1 + sound/soc/codecs/wm8962.c | 4 ++++ sound/soc/codecs/wm8996.c | 28 ++++++++++++++++++---------- sound/soc/ep93xx/ep93xx-i2s.c | 5 +++-- sound/soc/fsl/fsl_dma.c | 2 ++ sound/soc/fsl/mpc8610_hpcd.c | 18 +++++++++--------- sound/soc/fsl/p1022_ds.c | 4 +++- sound/soc/kirkwood/kirkwood-i2s.c | 2 +- sound/soc/samsung/Kconfig | 1 + sound/soc/samsung/speyside_wm8962.c | 6 ++++++ sound/soc/soc-core.c | 2 +- sound/soc/tegra/tegra_wm8903.c | 4 ++-- 12 files changed, 51 insertions(+), 26 deletions(-)

2 1

[PATCH] ALSA: hda: Conexant: Allow different output types to share DAC
by David Henningsson 25 Aug '11

25 Aug '11

Headphones has stopped working for the original reported (a regression compared to 2.6.38). This is because Speaker and Headphones share the same DAC, in which case no Headphones volume control was created. This patch fixes so that both Speaker and Headphones volume controls are created in such scenario. Buglink: http://bugs.launchpad.net/bugs/817943 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 40 ++++++++++++++++++++++------------------ 1 files changed, 22 insertions(+), 18 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 197ad93..a29b88d 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3327,22 +3327,23 @@ static int fill_cx_auto_dacs(struct hda_codec *codec, hda_nid_t *dacs) } /* fill pin_dac_pair list from the pin and dac list */ -static int fill_dacs_for_pins(struct hda_codec *codec, hda_nid_t *pins, +static void fill_dacs_for_pins(struct hda_codec *codec, hda_nid_t *pins, int num_pins, hda_nid_t *dacs, int *rest, - struct pin_dac_pair *filled, int type) + struct pin_dac_pair *filled, int *saved_nums, + int type) { int i, nums; - nums = 0; + nums = *saved_nums; for (i = 0; i < num_pins; i++) { filled[nums].pin = pins[i]; filled[nums].type = type; filled[nums].dac = get_unassigned_dac(codec, pins[i], dacs, rest); - if (!filled[nums].dac && i > 0 && filled[0].dac) + if (!filled[nums].dac && filled[0].dac) filled[nums].dac = filled[0].dac | DAC_SLAVE_FLAG; nums++; } - return nums; + *saved_nums = nums; } /* parse analog output paths */ @@ -3355,15 +3356,16 @@ static void cx_auto_parse_output(struct hda_codec *codec) rest = fill_cx_auto_dacs(codec, dacs); /* parse all analog output pins */ - nums = fill_dacs_for_pins(codec, cfg->line_out_pins, cfg->line_outs, - dacs, &rest, spec->dac_info, - AUTO_PIN_LINE_OUT); - nums += fill_dacs_for_pins(codec, cfg->hp_pins, cfg->hp_outs, - dacs, &rest, spec->dac_info + nums, - AUTO_PIN_HP_OUT); - nums += fill_dacs_for_pins(codec, cfg->speaker_pins, cfg->speaker_outs, - dacs, &rest, spec->dac_info + nums, - AUTO_PIN_SPEAKER_OUT); + nums = 0; + fill_dacs_for_pins(codec, cfg->line_out_pins, cfg->line_outs, + dacs, &rest, spec->dac_info, &nums, + AUTO_PIN_LINE_OUT); + fill_dacs_for_pins(codec, cfg->hp_pins, cfg->hp_outs, + dacs, &rest, spec->dac_info, &nums, + AUTO_PIN_HP_OUT); + fill_dacs_for_pins(codec, cfg->speaker_pins, cfg->speaker_outs, + dacs, &rest, spec->dac_info, &nums, + AUTO_PIN_SPEAKER_OUT); spec->dac_info_filled = nums; /* fill multiout struct */ for (i = 0; i < nums; i++) { @@ -4130,9 +4132,11 @@ static int try_add_pb_volume(struct hda_codec *codec, hda_nid_t dac, hda_nid_t pin, const char *name, int idx) { unsigned int caps; - caps = query_amp_caps(codec, dac, HDA_OUTPUT); - if (caps & AC_AMPCAP_NUM_STEPS) - return cx_auto_add_pb_volume(codec, dac, name, idx); + if (!(dac & DAC_SLAVE_FLAG)) { + caps = query_amp_caps(codec, dac, HDA_OUTPUT); + if (caps & AC_AMPCAP_NUM_STEPS) + return cx_auto_add_pb_volume(codec, dac, name, idx); + } caps = query_amp_caps(codec, pin, HDA_OUTPUT); if (caps & AC_AMPCAP_NUM_STEPS) return cx_auto_add_pb_volume(codec, pin, name, idx); @@ -4155,7 +4159,7 @@ static int cx_auto_build_output_controls(struct hda_codec *codec) const char *label; int idx, type; hda_nid_t dac = spec->dac_info[i].dac; - if (!dac || (dac & DAC_SLAVE_FLAG)) + if (!dac) continue; type = spec->dac_info[i].type; if (type == AUTO_PIN_LINE_OUT) -- 1.7.4.1 --------------060508080506000304070004--

1 0

[PATCH] ALSA: hda: Conexant: Allow different output types to share DAC
by David Henningsson 25 Aug '11

25 Aug '11

Headphones has stopped working for the original reported (a regression compared to 2.6.38). This is because Speaker and Headphones share the same DAC, in which case no Headphones volume control was created. This patch fixes so that both Speaker and Headphones volume controls are created in such scenario. BugLink: http://bugs.launchpad.net/bugs/817943 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 46 +++++++++++++++++++++++---------------- 1 files changed, 27 insertions(+), 19 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 197ad93..aa0e4b9 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3329,18 +3329,26 @@ static int fill_cx_auto_dacs(struct hda_codec *codec, hda_nid_t *dacs) /* fill pin_dac_pair list from the pin and dac list */ static int fill_dacs_for_pins(struct hda_codec *codec, hda_nid_t *pins, int num_pins, hda_nid_t *dacs, int *rest, - struct pin_dac_pair *filled, int type) + struct pin_dac_pair *filled, int nums, + int type) { - int i, nums; + int i, start = nums; - nums = 0; - for (i = 0; i < num_pins; i++) { + for (i = 0; i < num_pins; i++, nums++) { filled[nums].pin = pins[i]; filled[nums].type = type; filled[nums].dac = get_unassigned_dac(codec, pins[i], dacs, rest); - if (!filled[nums].dac && i > 0 && filled[0].dac) + if (filled[nums].dac) + continue; + if (filled[start].dac && get_connection_index(codec, pins[i], filled[start].dac) >= 0) { + filled[nums].dac = filled[start].dac | DAC_SLAVE_FLAG; + continue; + } + if (filled[0].dac && get_connection_index(codec, pins[i], filled[0].dac) >= 0) { filled[nums].dac = filled[0].dac | DAC_SLAVE_FLAG; - nums++; + continue; + } + snd_printdd("Failed to find a DAC for pin 0x%x", pins[i]); } return nums; } @@ -3356,14 +3364,14 @@ static void cx_auto_parse_output(struct hda_codec *codec) rest = fill_cx_auto_dacs(codec, dacs); /* parse all analog output pins */ nums = fill_dacs_for_pins(codec, cfg->line_out_pins, cfg->line_outs, - dacs, &rest, spec->dac_info, - AUTO_PIN_LINE_OUT); - nums += fill_dacs_for_pins(codec, cfg->hp_pins, cfg->hp_outs, - dacs, &rest, spec->dac_info + nums, - AUTO_PIN_HP_OUT); - nums += fill_dacs_for_pins(codec, cfg->speaker_pins, cfg->speaker_outs, - dacs, &rest, spec->dac_info + nums, - AUTO_PIN_SPEAKER_OUT); + dacs, &rest, spec->dac_info, 0, + AUTO_PIN_LINE_OUT); + nums = fill_dacs_for_pins(codec, cfg->hp_pins, cfg->hp_outs, + dacs, &rest, spec->dac_info, nums, + AUTO_PIN_HP_OUT); + nums = fill_dacs_for_pins(codec, cfg->speaker_pins, cfg->speaker_outs, + dacs, &rest, spec->dac_info, nums, + AUTO_PIN_SPEAKER_OUT); spec->dac_info_filled = nums; /* fill multiout struct */ for (i = 0; i < nums; i++) { @@ -4130,9 +4138,11 @@ static int try_add_pb_volume(struct hda_codec *codec, hda_nid_t dac, hda_nid_t pin, const char *name, int idx) { unsigned int caps; - caps = query_amp_caps(codec, dac, HDA_OUTPUT); - if (caps & AC_AMPCAP_NUM_STEPS) - return cx_auto_add_pb_volume(codec, dac, name, idx); + if (dac && !(dac & DAC_SLAVE_FLAG)) { + caps = query_amp_caps(codec, dac, HDA_OUTPUT); + if (caps & AC_AMPCAP_NUM_STEPS) + return cx_auto_add_pb_volume(codec, dac, name, idx); + } caps = query_amp_caps(codec, pin, HDA_OUTPUT); if (caps & AC_AMPCAP_NUM_STEPS) return cx_auto_add_pb_volume(codec, pin, name, idx); @@ -4155,8 +4165,6 @@ static int cx_auto_build_output_controls(struct hda_codec *codec) const char *label; int idx, type; hda_nid_t dac = spec->dac_info[i].dac; - if (!dac || (dac & DAC_SLAVE_FLAG)) - continue; type = spec->dac_info[i].type; if (type == AUTO_PIN_LINE_OUT) type = spec->autocfg.line_out_type; -- 1.7.4.1 --------------030805020208050803070801--

1 0

[alsa-devel] How to handle special SPI read/write for ASoC codec driver?
by Ashish Chavan 25 Aug '11

25 Aug '11

Hi, I am working on updating ASoC codec driver for Dialog's DA7210 codec (sound/soc/codec/da7210.c). I am planning to add SPI support and have a query. DA7210 codec has total 136, 8 bit registers (0x88). It supports 8 bit address and 8 bit data format for SPI. One out of eight address bits is used to indicate read/write operation, so effectively there are only seven bits to represent address. So, directly we can only address 128 (0 to 127) registers. In order to make other upper registers (128 to 136) addressable, DA7210 provides a special PAGE1 register addressed at (0x00 / 0x80). This PAGE1 register basically acts as an offset to normal registers addresses. If PAGE1 register's value is "0x0" then, register address "0" actually maps to 0th register. But if page register's value is "0x80", then register address "0" actually maps to 128th register, register "2" maps to 129th reg, and so on. While adding support for SPI, I need to add this logic of setting correct page while reading and writing. AFAIK, the proper place to add this extra logic is read/write functions on codec. i.e. something similar to, @@ -208,6 +208,38 @@ static int da7210_write(struct snd_soc_codec *codec, u32 reg, u32 value) BUG_ON(codec->driver->volatile_register); +#if defined(CONFIG_SPI_MASTER) + /* + * spi_page member of private data stores current status of page + * register. 0 => page 0 selected, 1 => page 1 selected + */ + struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); + if(reg >= 0x80){ + /* We are writing register in page 1 */ + if(!da7210->spi_page){ + /* Need to set page register first */ + data[0] = 0x00; + data[1] = 0x80; /* need to set seventh bit */ + if (2 != codec->hw_write(codec->control_data, data, 2)) + return -EIO; + cache[0] = 0x80; /* Update cache */ + da7210->spi_page = 1; /* Update local status */ + } + + }else { + /* We are writing register in page 0 */ + if(da7210->spi_page){ + /* Need to reset page register first */ + data[0] = 0x80; + data[1] = 0x00; /* need to reset seventh bit */ + if (2 != codec->hw_write(codec->control_data, data, 2)) + return -EIO; + cache[0x80] = 0x00; /* Update cache */ + da7210->spi_page = 0; /* Update local status */ + } + } +#endif + data[0] = reg & 0xff; data[1] = value & 0xff; Any comments on this code? Is there any better place to put this extra logic? Any way to accommodate this in to generic soc-io functions? Thanks, -- Ashish

1 0

[alsa-devel] How to handle special SPI read/write for ASoC codec driver?
by Ashish Chavan 25 Aug '11

25 Aug '11

Hi, I am working on updating ASoC codec driver for Dialog's DA7210 codec (sound/soc/codec/da7210.c). I am planning to add SPI support and have a query. DA7210 codec has total 136, 8 bit registers (0x88). It supports 8 bit address and 8 bit data format for SPI. One out of eight address bits is used to indicate read/write operation, so effectively there are only seven bits to represent address. So, directly we can only address 128 (0 to 127) registers. In order to make other upper registers (128 to 136) addressable, DA7210 provides a special PAGE1 register addressed at (0x00 / 0x80). This PAGE1 register basically acts as an offset to normal registers addresses. If PAGE1 register's value is "0x0" then, register address "0" actually maps to 0th register. But if page register's value is "0x80", then register address "0" actually maps to 128th register, register "2" maps to 129th reg, and so on. While adding support for SPI, I need to add this logic of setting correct page while reading and writing. AFAIK, the proper place to add this extra logic is read/write functions on codec. i.e. something similar to, @@ -208,6 +208,38 @@ static int da7210_write(struct snd_soc_codec *codec, u32 reg, u32 value) BUG_ON(codec->driver->volatile_register); +#if defined(CONFIG_SPI_MASTER) + /* + * spi_page member of private data stores current status of page + * register. 0 => page 0 selected, 1 => page 1 selected + */ + struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); + if(reg >= 80){ + /* We are writing register in page 1 */ + if(!da7210->spi_page){ + /* Need to set page register first */ + data[0] = 0x00; + data[1] = 0x80; /* need to set seventh bit */ + if (2 != codec->hw_write(codec->control_data, data, 2)) + return -EIO; + cache[0] = 0x80; /* Update cache */ + da7210->spi_page = 1; /* Update local status */ + } + + }else { + /* We are writing register in page 0 */ + if(da7210->spi_page){ + /* Need to reset page register first */ + data[0] = 0x80; + data[1] = 0x00; /* need to reset seventh bit */ + if (2 != codec->hw_write(codec->control_data, data, 2)) + return -EIO; + cache[80] = 0x00; /* Update cache */ + da7210->spi_page = 0; /* Update local status */ + } + } +#endif + data[0] = reg & 0xff; data[1] = value & 0xff; Any comments on this code? Is there any better place to put this extra logic? Any way to accommodate this in to generic soc-io functions? Thanks, -- Ashish

1 0

[alsa-devel] [PATCH] ASoC: MPC5200: replace of_device with platform_device
by Timur Tabi 25 Aug '11

25 Aug '11

'struct of_device' no longer exists, and its functionality has been merged into platform_device. Update the MPC5200 audio DMA driver (mpc5200_dma) accordingly. This fixes a build break. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/mpc5200_dma.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index fd0dc46..5c6c245 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -369,7 +369,7 @@ static struct snd_soc_platform_driver mpc5200_audio_dma_platform = { .pcm_free = &psc_dma_free, }; -static int mpc5200_hpcd_probe(struct of_device *op) +static int mpc5200_hpcd_probe(struct platform_device *op) { phys_addr_t fifo; struct psc_dma *psc_dma; @@ -487,7 +487,7 @@ out_unmap: return ret; } -static int mpc5200_hpcd_remove(struct of_device *op) +static int mpc5200_hpcd_remove(struct platform_device *op) { struct psc_dma *psc_dma = dev_get_drvdata(&op->dev); @@ -519,7 +519,7 @@ MODULE_DEVICE_TABLE(of, mpc5200_hpcd_match); static struct platform_driver mpc5200_hpcd_of_driver = { .probe = mpc5200_hpcd_probe, .remove = mpc5200_hpcd_remove, - .dev = { + .driver = { .owner = THIS_MODULE, .name = "mpc5200-pcm-audio", .of_match_table = mpc5200_hpcd_match, -- 1.7.3.4

2 7

[alsa-devel] Conexant 5051 lenovo-x200 fixes
by Ulo Mets 24 Aug '11

24 Aug '11

Hi, I am not a kernel developer, but occasionally dig into it if something gets in my way. On Lenovo X200 Tablet none of the mixers showed Mic Boost adjustments, but HDA-Analyzer found them and adjusted properly. So I inserted the corresponding controls, and it worked. Then, during testing I found that the labels of External and Docking Mic are swapped, so I exchanged these too. Since I could not test it on other hardware, I put it all on lenovo-x200 branch. The patch applies to 3.1-rc2, I hope it is sufficiently recent. The Mic Switches don't have any effect on my unit (X200T without modem) With best regards, Ulo Mets Signed-off-by: Ulo Mets<umets at osa.pri.ee> ---b/sound/pci/hda/patch_conexant.c 2011-08-01 12:45:32.302640276 +0300 +++ a/sound/pci/hda/patch_conexant.c 2011-08-01 12:50:22.003141176 +0300 @@ -1785,6 +1785,19 @@ {} }; +static const struct snd_kcontrol_new cxt5051_lenovo_x200_mixers[] = { + HDA_CODEC_VOLUME("Int Mic Volume", 0x14, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Int Mic Switch", 0x14, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Dock Mic Volume", 0x14, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Dock Mic Switch", 0x14, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Ext Mic Volume", 0x15, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Ext Mic Switch", 0x15, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Boost", 0x1d, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Dock Mic Boost", 0x17, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Ext Mic Boost", 0x18, 0x00, HDA_INPUT), + {} +}; + static const struct snd_kcontrol_new cxt5051_hp_mixers[] = { HDA_CODEC_VOLUME("Internal Mic Volume", 0x14, 0x00, HDA_INPUT), HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT), @@ -2055,6 +2079,7 @@ break; case CXT5051_LENOVO_X200: spec->init_verbs[0] = cxt5051_lenovo_x200_init_verbs; + spec->mixers[0] = cxt5051_lenovo_x200_mixers; /* Thinkpad X301 does not have S/PDIF wired and no ability to use a docking station. */ if (codec->subsystem_id == 0x17aa211f)

2 14

[alsa-devel] [PATCH] ASoC: Correct element count for WM8996 sidetone HPF
by Mark Brown 24 Aug '11

24 Aug '11

I can count. Honest. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 73a1588..3c9080f 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -420,7 +420,7 @@ static const char *sidetone_hpf_text[] = { }; static const struct soc_enum sidetone_hpf = - SOC_ENUM_SINGLE(WM8996_SIDETONE, 7, 6, sidetone_hpf_text); + SOC_ENUM_SINGLE(WM8996_SIDETONE, 7, 7, sidetone_hpf_text); static const char *hpf_mode_text[] = { "HiFi", "Custom", "Voice" -- 1.7.5.4

2 1

[alsa-devel] [PATCH 1/3] ALSA: core: trivial code style fix
by Lu Guanqun 24 Aug '11

24 Aug '11

remove trailing tab on the line. Signed-off-by: Lu Guanqun <guanqun.lu(a)intel.com> --- sound/core/control.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index f8c5be4..7f2b3a7 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1072,7 +1072,7 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, long private_size; struct user_element *ue; int idx, err; - + if (card->user_ctl_count >= MAX_USER_CONTROLS) return -ENOMEM; if (info->count < 1)

2 16

[alsa-devel] [PATCH 1/2] ASoC: mxs-saif: add record function
by Dong Aisheng 24 Aug '11

24 Aug '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> --- include/sound/saif.h | 16 +++++ sound/soc/mxs/mxs-saif.c | 162 ++++++++++++++++++++++++++++++++++++++++++---- sound/soc/mxs/mxs-saif.h | 4 + 3 files changed, 168 insertions(+), 14 deletions(-) diff --git a/include/sound/saif.h b/include/sound/saif.h new file mode 100644 index 0000000..7a98d9b --- /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 __MACH_MXS_SAIF_H__ +#define __MACH_MXS_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..eb8ba79 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -27,15 +27,39 @@ #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 +76,36 @@ 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 +114,27 @@ 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 +148,29 @@ 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 +212,7 @@ 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 +258,7 @@ 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 +271,12 @@ 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 +360,21 @@ 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 +489,11 @@ 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; + + master_saif = mxs_saif_get_master(saif); + if (!master_saif) + return -EINVAL; switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -403,10 +501,20 @@ 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 +530,37 @@ 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) + if (!master_saif->mclk_in_use) { + __raw_writel(BM_SAIF_CTRL_RUN, + master_saif->base + SAIF_CTRL + MXS_CLR_ADDR); + udelay(100); + } + clk_disable(master_saif->clk); + + if (saif != master_saif) { __raw_writel(BM_SAIF_CTRL_RUN, saif->base + SAIF_CTRL + MXS_CLR_ADDR); + /* wait a while for the current sample to complete */ + udelay(100); + clk_disable(saif->clk); + } + + master_saif->ongoing = 0; break; default: @@ -519,15 +644,24 @@ 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

4 4

[alsa-devel] [PATCH 1/1] ASoC: soc-core: check rate for symmetry only when pcm is ongoing
by Dong Aisheng 24 Aug '11

24 Aug '11

For the playback and record using different dai links, checking !rtd->rate for symmetry may not be accurate because that pcm may be acutually not running and the default new open rate is 0, then the warning message "Not enforcing symmetric_rates" will happen each time with running arecord | aplay. Now we only check rate for symmetry when the pcm is really ongoing which seems more sensible. 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> --- include/sound/soc.h | 1 + sound/soc/soc-pcm.c | 5 +++++ 2 files changed, 6 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 3fe658e..8e9931c 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -851,6 +851,7 @@ struct snd_soc_pcm_runtime { /* Symmetry data - only valid if symmetry is being enforced */ unsigned int rate; + unsigned int ongoing; long pmdown_time; /* runtime devices */ diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 1aee9fc..f3c1a66 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -39,6 +39,9 @@ static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream) !rtd->dai_link->symmetric_rates) return 0; + if (!rtd->ongoing) + return 0; + /* This can happen if multiple streams are starting simultaneously - * the second can need to get its constraints before the first has * picked a rate. Complain and allow the application to carry on. @@ -209,6 +212,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) cpu_dai->active++; codec_dai->active++; rtd->codec->active++; + rtd->ongoing = 1; mutex_unlock(&rtd->pcm_mutex); return 0; @@ -322,6 +326,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) SND_SOC_DAPM_STREAM_STOP); } + rtd->ongoing = 0; mutex_unlock(&rtd->pcm_mutex); return 0; } -- 1.7.0.4

4 5

[alsa-devel] [PATCH] ASoC: SAMSUNG: Add Kconfig to support SMDK4212
by Sangbeom Kim 24 Aug '11

24 Aug '11

This patch adds Kconfig to support SMDK4212. SMDK4212 is based on samsung exynos4212 SoC. And WM8994 is used for audio codec. Signed-off-by: Sangbeom Kim <sbkim73(a)samsung.com> --- sound/soc/samsung/Kconfig | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index b99091f..9a3138b 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -63,7 +63,7 @@ config SND_SOC_SAMSUNG_SMDK_WM8580 config SND_SOC_SAMSUNG_SMDK_WM8994 tristate "SoC I2S Audio support for WM8994 on SMDK" - depends on SND_SOC_SAMSUNG && (MACH_SMDKV310 || MACH_SMDKC210) + depends on SND_SOC_SAMSUNG && (MACH_SMDKV310 || MACH_SMDKC210 || MACH_SMDK4212) select SND_SOC_WM8994 select SND_SAMSUNG_I2S help @@ -158,7 +158,7 @@ config SND_SOC_GONI_AQUILA_WM8994 config SND_SOC_SAMSUNG_SMDK_SPDIF tristate "SoC S/PDIF Audio support for SMDK" - depends on SND_SOC_SAMSUNG && (MACH_SMDKC100 || MACH_SMDKC110 || MACH_SMDKV210 || MACH_SMDKV310) + depends on SND_SOC_SAMSUNG && (MACH_SMDKC100 || MACH_SMDKC110 || MACH_SMDKV210 || MACH_SMDKV310 || MACH_SMDK4212) select SND_SAMSUNG_SPDIF help Say Y if you want to add support for SoC S/PDIF audio on the SMDK. @@ -173,7 +173,7 @@ config SND_SOC_SMDK_WM8580_PCM config SND_SOC_SMDK_WM8994_PCM tristate "SoC PCM Audio support for WM8994 on SMDK" - depends on SND_SOC_SAMSUNG && (MACH_SMDKC210 || MACH_SMDKV310) + depends on SND_SOC_SAMSUNG && (MACH_SMDKC210 || MACH_SMDKV310 || MACH_SMDK4212) select SND_SOC_WM8994 select SND_SAMSUNG_PCM help -- 1.7.1

3 2

[alsa-devel] [PATCH] ASoC: Tegra: wm8903 machine driver: Drop Ventana support
by Stephen Warren 24 Aug '11

24 Aug '11

Board file support for Ventana is not yet mainlined, and probably won't ever be given the move to Device-Tree. Consequently, the Ventana entry is being removed from arch/arm/tools/mach-types in the next merge window, since it was registered over a year ago. This will also remove function machine_is_ventana(), which is used by the ASoC Tegra WM8903 machine driver. This will cause compilation failures. Drop Ventana support to resolve this. Hopefully, in the not-too-distant future, tegra_wm8903.c will be able to configure itself from Device-Tree, and hence we'll be able to re-instate Ventana support just by creating a .dts file for the board. Also note that Aebl support is in a similar boat. However, that board isn't scheduled for deprecation for at least another 5 months, and perhaps we will have completely removed non-Device-Tree support from tegra_wm8903.c by then and/or adjusted mach-types policy. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/tegra_wm8903.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 661373c..be27f1d 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -319,7 +319,7 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_force_enable_pin(dapm, "Mic Bias"); /* FIXME: Calculate automatically based on DAPM routes? */ - if (!machine_is_harmony() && !machine_is_ventana()) + if (!machine_is_harmony()) snd_soc_dapm_nc_pin(dapm, "IN1L"); if (!machine_is_seaboard() && !machine_is_aebl()) snd_soc_dapm_nc_pin(dapm, "IN1R"); @@ -395,7 +395,7 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); - if (machine_is_harmony() || machine_is_ventana()) { + if (machine_is_harmony()) { card->dapm_routes = harmony_audio_map; card->num_dapm_routes = ARRAY_SIZE(harmony_audio_map); } else if (machine_is_seaboard()) { -- 1.7.0.4

3 2

Re: [alsa-devel] [linux-next] ASoC: build failure if CONFIG_I2C=m && CONFIG_SPI=y && CONFIG_SND_SOC=y
by Randy Dunlap 24 Aug '11

24 Aug '11

On Tue, 23 Aug 2011 19:46:00 +0800 Axel Lin wrote: > > hi, > I got below build failure if CONFIG_I2C=m && CONFIG_SPI=y && CONFIG_SND_SOC=y > > LD .tmp_vmlinux1 > sound/built-in.o: In function `snd_soc_codec_set_cache_io': > last.c:(.text+0x1cb80): undefined reference to `regmap_init_i2c' > make: *** [.tmp_vmlinux1] Error 1 I don't know if this is the same failure or just similar: ERROR: "regmap_exit" [drivers/mfd/wm8400-core.ko] undefined! ERROR: "regmap_bulk_read" [drivers/mfd/wm8400-core.ko] undefined! ERROR: "regmap_read" [drivers/mfd/wm8400-core.ko] undefined! ERROR: "regmap_init_i2c" [drivers/mfd/wm8400-core.ko] undefined! ERROR: "regmap_raw_read" [drivers/mfd/wm8400-core.ko] undefined! ERROR: "regmap_write" [drivers/mfd/wm8400-core.ko] undefined! Maybe MFD_WM8400 needs to select REGMAP_I2C ? --- ~Randy *** Remember to use Documentation/SubmitChecklist when testing your code ***

2 1

[alsa-devel] [PATCH] ASoC: Add samsung maintainer
by Sangbeom Kim 24 Aug '11

24 Aug '11

Cc: Jassi Brar <jassisinghbrar(a)gmail.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg(a)ti.com> Signed-off-by: Sangbeom Kim <sbkim73(a)samsung.com> --- MAINTAINERS | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 069ee3b..fadff23 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -5532,6 +5532,7 @@ F: include/media/*7146* SAMSUNG AUDIO (ASoC) DRIVERS M: Jassi Brar <jassisinghbrar(a)gmail.com> +M: Sangbeom Kim <sbkim73(a)samsung.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Supported F: sound/soc/samsung -- 1.7.1

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Re: [alsa-devel] [PATCH 1/2] ASoC: Add Springbank I/O card to Speyside Kconfig
by Mark Brown 24 Aug '11

24 Aug '11

On Tue, Aug 23, 2011 at 10:41:50AM +0100, Joseph Pentland wrote: > Signed-off-by: Joseph Pentland <jp(a)opensource.wolfsonmicro.com> Applied, thanks.

1 0

Re: [alsa-devel] [PATCH] ALSA: ctxfi: Bump playback substreams to 256
by Takashi Iwai 24 Aug '11

24 Aug '11

At Wed, 24 Aug 2011 00:48:59 +0200, Maarten Lankhorst wrote: > > There are references in the code to 256 sources, so I tested it with 256 aplays, > of which the first and last with real data and the rest playing /dev/zero . > > Also increase amount of page tables, so the default aplay size works. > > Signed-off-by: Maarten Lankhorst <m.b.lankhorst(a)gmail.com> Applied now. Thanks. Takashi > --- > diff --git a/sound/pci/ctxfi/ctpcm.c b/sound/pci/ctxfi/ctpcm.c > index 457d211..2c86226 100644 > --- a/sound/pci/ctxfi/ctpcm.c > +++ b/sound/pci/ctxfi/ctpcm.c > @@ -404,7 +404,7 @@ int ct_alsa_pcm_create(struct ct_atc *atc, > int err; > int playback_count, capture_count; > > - playback_count = (IEC958 == device) ? 1 : 8; > + playback_count = (IEC958 == device) ? 1 : 256; > capture_count = (FRONT == device) ? 1 : 0; > err = snd_pcm_new(atc->card, "ctxfi", device, > playback_count, capture_count, &pcm); > diff --git a/sound/pci/ctxfi/ctsrc.c b/sound/pci/ctxfi/ctsrc.c > index c749fa7..e134b3a 100644 > --- a/sound/pci/ctxfi/ctsrc.c > +++ b/sound/pci/ctxfi/ctsrc.c > @@ -20,7 +20,7 @@ > #include "cthardware.h" > #include <linux/slab.h> > > -#define SRC_RESOURCE_NUM 64 > +#define SRC_RESOURCE_NUM 256 > #define SRCIMP_RESOURCE_NUM 256 > > static unsigned int conj_mask; > diff --git a/sound/pci/ctxfi/ctvmem.h b/sound/pci/ctxfi/ctvmem.h > index b23adfc..e6da60e 100644 > --- a/sound/pci/ctxfi/ctvmem.h > +++ b/sound/pci/ctxfi/ctvmem.h > @@ -18,7 +18,7 @@ > #ifndef CTVMEM_H > #define CTVMEM_H > > -#define CT_PTP_NUM 1 /* num of device page table pages */ > +#define CT_PTP_NUM 4 /* num of device page table pages */ > > #include <linux/mutex.h> > #include <linux/list.h> >

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Re: [alsa-devel] [PATCH 2/2] USB: Gadget: Add Audio Class 2.0 Driver
by Jassi Brar 24 Aug '11

24 Aug '11

On Wed, Aug 24, 2011 at 7:30 AM, Chen Peter-B29397 <B29397(a)freescale.com> wrote: >> >> This is a flexible USB Audio Class 2.0 compliant gadget driver that >> implements a simple topology with a virtual sound card exposed at >> the function side. >> >> The driver doesn't expect any real audio codec to be present on the >> function - the audio streams are simply sinked to and sourced from a >> virtual ALSA sound card created. The user-space application may choose >> to do whatever it wants with the data received from the USB Host and >> choose to provide whatever it wants as audio data to the USB Host. >> >> Capture(USB-Out) and Playback(USB-In) can be run at independent >> configurations specified via module parameters while loading the driver. >> > > Have you a chance to test at Macbook (10.6.5+) which has USB Audio 2.0 support well? > Nopes, haven't had my hands on one yet. It would help if someone could verify. ..... >> +/* STD AS ISO OUT Endpoint */ >> +struct usb_endpoint_descriptor fs_epout_desc = { >> + .bLength = USB_DT_ENDPOINT_SIZE, >> + .bDescriptorType = USB_DT_ENDPOINT, >> + >> + .bEndpointAddress = USB_DIR_OUT, >> + .bmAttributes = USB_ENDPOINT_XFER_ISOC | USB_ENDPOINT_SYNC_ASYNC, >> + .bInterval = 1, >> +}; >> + >> +struct usb_endpoint_descriptor hs_epout_desc = { >> + .bLength = USB_DT_ENDPOINT_SIZE, >> + .bDescriptorType = USB_DT_ENDPOINT, >> + >> + .bmAttributes = USB_ENDPOINT_XFER_ISOC | USB_ENDPOINT_SYNC_ASYNC, >> + .bInterval = 4, >> +}; > The sync type is asynchronous, do you have plan to add feedback for handling > overflow and underflow? > umm... I was not sure if the feature is worth the overhead/complexity of maintaining timers in the driver. Though I welcome hearing why not so?

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Re: [alsa-devel] [PATCH 1/2] ASoC: Add Springbank I/O card to Speyside Kconfig
by Liam Girdwood 23 Aug '11

23 Aug '11

On 23/08/11 10:41, Joseph Pentland wrote: > Signed-off-by: Joseph Pentland <jp(a)opensource.wolfsonmicro.com> > --- > sound/soc/samsung/Kconfig | 1 + > 1 files changed, 1 insertions(+), 0 deletions(-) > > diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig > index b99091f..65f980e 100644 > --- a/sound/soc/samsung/Kconfig > +++ b/sound/soc/samsung/Kconfig > @@ -185,6 +185,7 @@ config SND_SOC_SPEYSIDE > select SND_SAMSUNG_I2S > select SND_SOC_WM8996 > select SND_SOC_WM9081 > + select SND_SOC_WM1250_EV1 > > config SND_SOC_SPEYSIDE_WM8962 > tristate "Audio support for Wolfson Speyside with WM8962" Acked-by: Liam Girdwood <lrg(a)ti.com> Missing 2/2 ?

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Re: [alsa-devel] Headset driver detection problem during bootup
by anish singh 23 Aug '11

23 Aug '11

adding more information and experts on alsa. On Fri, Aug 19, 2011 at 9:23 PM, anish singh <anish198519851985(a)gmail.com> wrote: > I need some suggestions for below situation: > > Headset if it is inserted before the device is powered up is > now detected by forcefully calling the interrupt handler in > the probe function of the headset driver. Headset is detected in our andorid device by using a detect gpio which is turned on when headset is inserted.If the headset is inserted right from very beginging i.e. before device is even turned on then we forcefully call the handler and thereby in handler we check if gpio is turned on or not and do the rest of things. > Once the handler is called then we find out if the headset > is there or not using gpio and then normal detection happens > > Is it the right way to detect the device during power-up? Does not linux have any other way of doing this which i may be missing? How this is handled in the case of ALSA jack i.e. the jack drivers which is using alsa framework for detection? Android doesn't use ALSA framework for reporting JACK. > > After the device is powered up and then if we insert headset > then it is working fine by calling the interrupt handler and > everything goes fine. > > linux -2.3.5 > Android Device >

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[PATCH] ALSA: hda: hdmi: Hint matching between input devices and pcm devices
by David Henningsson 23 Aug '11

23 Aug '11

Since modern HDMI cards often have more than one output pin and thus input device, we need to know which one has actually been plugged in. This patch adds a name hint that indicates which PCM device is connected to which pin. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_hdmi.c | 31 ++++++++++++++++++++++++------- 1 files changed, 24 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 19cb72d..8c83ec4 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -967,19 +967,12 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) per_pin->pin_nid = pin_nid; - err = snd_hda_input_jack_add(codec, pin_nid, - SND_JACK_VIDEOOUT, NULL); - if (err < 0) - return err; - err = hdmi_read_pin_conn(codec, pin_idx); if (err < 0) return err; spec->num_pins++; - hdmi_present_sense(codec, pin_nid, eld); - return 0; } @@ -1162,6 +1155,25 @@ static int generic_hdmi_build_pcms(struct hda_codec *codec) return 0; } +static int generic_hdmi_build_jack(struct hda_codec *codec, int pin_idx) +{ + int err; + char hdmi_str[32]; + struct hdmi_spec *spec = codec->spec; + struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx]; + int pcmdev = spec->pcm_rec[pin_idx].device; + + snprintf(hdmi_str, sizeof(hdmi_str), "HDMI/DP (pcm %d)", pcmdev); + + err = snd_hda_input_jack_add(codec, per_pin->pin_nid, + SND_JACK_VIDEOOUT, pcmdev > 0 ? hdmi_str : NULL); + if (err < 0) + return err; + + hdmi_present_sense(codec, per_pin->pin_nid, &per_pin->sink_eld); + return 0; +} + static int generic_hdmi_build_controls(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; @@ -1170,6 +1182,11 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx]; + + err = generic_hdmi_build_jack(codec, pin_idx); + if (err < 0) + return err; + err = snd_hda_create_spdif_out_ctls(codec, per_pin->pin_nid, per_pin->mux_nids[0]); -- 1.7.4.1 --------------040603040001050104040801--

1 0

[PATCH] ALSA: hda: hdmi: Hint matching between input devices and pcm devices
by David Henningsson 23 Aug '11

23 Aug '11

Since modern HDMI cards often have more than one output pin and thus input device, we need to know which one has actually been plugged in. This patch adds a name hint that indicates which PCM device is connected to which pin. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_hdmi.c | 31 ++++++++++++++++++++++++------- 1 files changed, 24 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 19cb72d..8c83ec4 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -967,19 +967,12 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) per_pin->pin_nid = pin_nid; - err = snd_hda_input_jack_add(codec, pin_nid, - SND_JACK_VIDEOOUT, NULL); - if (err < 0) - return err; - err = hdmi_read_pin_conn(codec, pin_idx); if (err < 0) return err; spec->num_pins++; - hdmi_present_sense(codec, pin_nid, eld); - return 0; } @@ -1162,6 +1155,25 @@ static int generic_hdmi_build_pcms(struct hda_codec *codec) return 0; } +static int generic_hdmi_build_jack(struct hda_codec *codec, int pin_idx) +{ + int err; + char hdmi_str[32]; + struct hdmi_spec *spec = codec->spec; + struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx]; + int pcmdev = spec->pcm_rec[pin_idx].device; + + snprintf(hdmi_str, sizeof(hdmi_str), "HDMI/DP,pcm=%d", pcmdev); + + err = snd_hda_input_jack_add(codec, per_pin->pin_nid, + SND_JACK_VIDEOOUT, pcmdev > 0 ? hdmi_str : NULL); + if (err < 0) + return err; + + hdmi_present_sense(codec, per_pin->pin_nid, &per_pin->sink_eld); + return 0; +} + static int generic_hdmi_build_controls(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; @@ -1170,6 +1182,11 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx]; + + err = generic_hdmi_build_jack(codec, pin_idx); + if (err < 0) + return err; + err = snd_hda_create_spdif_out_ctls(codec, per_pin->pin_nid, per_pin->mux_nids[0]); -- 1.7.4.1 --------------010607010305070404080608--

1 0

[PATCH] ALSA: hda: hdmi: Hint matching between input devices and pcm devices
by David Henningsson 23 Aug '11

23 Aug '11

Since modern HDMI cards often have more than one output pin and thus input device, we need to know which one has actually been plugged in. This patch adds a name hint that indicates which PCM device is connected to which pin. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_hdmi.c | 31 ++++++++++++++++++++++++------- 1 files changed, 24 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 19cb72d..8c83ec4 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -967,19 +967,12 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) per_pin->pin_nid = pin_nid; - err = snd_hda_input_jack_add(codec, pin_nid, - SND_JACK_VIDEOOUT, NULL); - if (err < 0) - return err; - err = hdmi_read_pin_conn(codec, pin_idx); if (err < 0) return err; spec->num_pins++; - hdmi_present_sense(codec, pin_nid, eld); - return 0; } @@ -1162,6 +1155,25 @@ static int generic_hdmi_build_pcms(struct hda_codec *codec) return 0; } +static int generic_hdmi_build_jack(struct hda_codec *codec, int pin_idx) +{ + int err; + char hdmi_str[32]; + struct hdmi_spec *spec = codec->spec; + struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx]; + int pcmdev = spec->pcm_rec[pin_idx].device; + + snprintf(hdmi_str, sizeof(hdmi_str), "HDMI/DP,pcm=%d", pcmdev); + + err = snd_hda_input_jack_add(codec, per_pin->pin_nid, + SND_JACK_VIDEOOUT, pcmdev > 0 ? hdmi_str : NULL); + if (err < 0) + return err; + + hdmi_present_sense(codec, per_pin->pin_nid, &per_pin->sink_eld); + return 0; +} + static int generic_hdmi_build_controls(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; @@ -1170,6 +1182,11 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx]; + + err = generic_hdmi_build_jack(codec, pin_idx); + if (err < 0) + return err; + err = snd_hda_create_spdif_out_ctls(codec, per_pin->pin_nid, per_pin->mux_nids[0]); -- 1.7.4.1 --------------040904060507060402050704--

1 0

[alsa-devel] ALSA calls don't work within Mozilla plug-ins
by Olivier Guillion - Myriad 23 Aug '11

23 Aug '11

Hi, I'm trying to develop a Mozilla plug-in that display and plays music scores. The music rendering module uses ALSA for sound (pcm) output. It seems that, when called from a Mozilla plug-in, any ALSA function that gets a pointer as parameter either crashes or returns wrong values. I made a very simple test plug-in from the "Basic Plugin" source code provided by Mozilla (I can provide the full project if needed). Here is what happens: - Calling "snd_asoundlib_version" works - Calling "snd_device_name_hint" crashes - If the browser runs in super-user mode (sudo firefox), all works fine. - If I load the plug-in (which is a shared library) from a program of my own and call the test function, it works. Does anyone have any idea about what could cause this issue? Could it be related to access rights or memory allocation/mapping? I'm using: Ubuntu 11.04 Firefox 6.0 ALSA 1.0.24.1 I'm really stuck here, any advice would be highly appreciated. Thank you, Olivier Olivier Guillion Myriad 26 rue Michel de Montaigne 31200 Toulouse FRANCE -------- WebSite ----------- http://www.myriad-online.com Download here the latest version of our shareware programs, view the online tutorials or get info about Myriad and its activities. ---------------------------------

1 1

[PATCH] alsa-plugins: Pulse: only underrun if no more data has been written
by David Henningsson 23 Aug '11

23 Aug '11

If more data has already been written after the underrun, the underrun will automatically end and therefore we should not report it or restart the stream. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- pulse/pcm_pulse.c | 36 ++++++++++++++++++++++++++++++++++-- 1 files changed, 34 insertions(+), 2 deletions(-) diff --git a/pulse/pcm_pulse.c b/pulse/pcm_pulse.c index d6c6792..92f4777 100644 --- a/pulse/pcm_pulse.c +++ b/pulse/pcm_pulse.c @@ -40,8 +40,10 @@ typedef struct snd_pcm_pulse { size_t ptr; int underrun; int handle_underrun; + int handle_underrun_detect; size_t offset; + int64_t written; pa_stream *stream; @@ -460,6 +462,7 @@ static snd_pcm_sframes_t pulse_write(snd_pcm_ioplug_t * io, /* Make sure the buffer pointer is in sync */ pcm->last_size -= writebytes; + pcm->written += writebytes; ret = update_ptr(pcm); if (ret < 0) goto finish; @@ -585,6 +588,23 @@ static void stream_request_cb(pa_stream * p, size_t length, void *userdata) update_active(pcm); } +#if defined(PA_CHECK_VERSION) && PA_CHECK_VERSION(0,99,0) + +#define has_underrun_detect 1 + +static inline int do_underrun_detect(snd_pcm_pulse_t *pcm, pa_stream *p) +{ + return pcm->handle_underrun_detect && + pcm->written <= pa_stream_get_underflow_index(p); +} + +#else + +#define has_underrun_detect 0 +#define do_underrun_detect(pcm, p) 0 + +#endif + static void stream_underrun_cb(pa_stream * p, void *userdata) { snd_pcm_pulse_t *pcm = userdata; @@ -594,7 +614,8 @@ static void stream_underrun_cb(pa_stream * p, void *userdata) if (!pcm->p) return; - pcm->underrun = 1; + if (pcm->handle_underrun || do_underrun_detect(pcm, p)) + pcm->underrun = 1; } static void stream_latency_cb(pa_stream *p, void *userdata) { @@ -697,7 +718,7 @@ static int pulse_prepare(snd_pcm_ioplug_t * io) if (io->stream == SND_PCM_STREAM_PLAYBACK) { pa_stream_set_write_callback(pcm->stream, stream_request_cb, pcm); - if (pcm->handle_underrun) + if (pcm->handle_underrun_detect || pcm->handle_underrun) pa_stream_set_underflow_callback(pcm->stream, stream_underrun_cb, pcm); r = pa_stream_connect_playback(pcm->stream, pcm->device, @@ -739,6 +760,7 @@ static int pulse_prepare(snd_pcm_ioplug_t * io) pcm->offset = 0; pcm->underrun = 0; + pcm->written = 0; /* Reset fake ringbuffer */ pcm->last_size = 0; @@ -984,6 +1006,7 @@ SND_PCM_PLUGIN_DEFINE_FUNC(pulse) const char *device = NULL; const char *fallback_name = NULL; int handle_underrun = 0; + int handle_underrun_detect = 1; int err; snd_pcm_pulse_t *pcm; @@ -1017,6 +1040,14 @@ SND_PCM_PLUGIN_DEFINE_FUNC(pulse) handle_underrun = err; continue; } + if (strcmp(id, "handle_underrun_detect") == 0) { + if ((err = snd_config_get_bool(n)) < 0) { + SNDERR("Invalid value for %s", id); + return -EINVAL; + } + handle_underrun_detect = err; + continue; + } if (strcmp(id, "fallback") == 0) { if (snd_config_get_string(n, &fallback_name) < 0) { SNDERR("Invalid value for %s", id); @@ -1051,6 +1082,7 @@ SND_PCM_PLUGIN_DEFINE_FUNC(pulse) } pcm->handle_underrun = handle_underrun; + pcm->handle_underrun_detect = has_underrun_detect && handle_underrun_detect; err = pulse_connect(pcm->p, server, fallback_name != NULL); if (err < 0) -- 1.7.5.4 --------------040808080001070501040605--

1 0

[alsa-devel] [RME MADI/RayDAT/AIO/AES] Changing hdspm_config and hdspm_status
by Adrian Knoth 23 Aug '11

23 Aug '11

Hi! Florian Faber has sent me some new bits of his work on hdspm. While the changes to hdspm.c are straight-forward, he also cleaned up struct hdspm_config and hdspm_status, thus changing the userspace API/ABI: diff --git a/include/sound/hdspm.h b/include/sound/hdspm.h index 1f59ea2..25eab09 100644 --- a/include/sound/hdspm.h +++ b/include/sound/hdspm.h @@ -58,17 +58,20 @@ struct hdspm_peak_rms { /* ------------ CONFIG block IOCTL ---------------------- */ struct hdspm_config { - unsigned char pref_sync_ref; - unsigned char wordclock_sync_check; - unsigned char madi_sync_check; - unsigned int system_sample_rate; - unsigned int autosync_sample_rate; - unsigned char system_clock_mode; - unsigned char clock_source; - unsigned char autosync_ref; - unsigned char line_out; - unsigned int passthru; - unsigned int analog_out; + uint32_t dds; /* dds increment */ + uint16_t freq; /* 32000/44100/48000 */ + uint8_t wck_mul; /* 1/2/4 */ + uint8_t clock_mode; /* 0=master, 1=slave */ + union { + struct { + uint8_t line_out; /* 0=disabled, 1=enabled */ + uint8_t tms; /* 0=disabled, 1=enabled */ + uint8_t input_select; /* 0=BNC, 1=optical */ + uint8_t output_mode; /* 0=56ch, 1=64ch */ + uint8_t output_frame; /* 0=48K, 1=96K */ + uint8_t sync_ref; /* 0=WCK, 1=MADI, 2=TCO/SYNC_IN, 3=SYNC_IN */ + } madi; + } card_specific; }; #define SNDRV_HDSPM_IOCTL_GET_CONFIG \ @@ -161,7 +164,6 @@ struct hdspm_status { uint8_t sync_madi; /* enum hdspm_sync */ uint8_t sync_tco; /* enum hdspm_sync */ uint8_t sync_in; /* enum hdspm_sync */ - uint8_t madi_input; /* enum hdspm_madi_input */ uint8_t channel_format; /* enum hdspm_madi_channel_format */ uint8_t frame_format; /* enum hdspm_madi_frame_format */ } madi; I guess this means we'd need to redefine two ioctl numbers. Besides this, are there any current users of this code? The changes are not upstream, yet, so we could arrange for a grace period, maybe saying something like "Starting with Linux-3.x and ALSA-1.0.y, you'll have to change your code". X and Y are yet to be chosen wisely. If you're interested to check if the upcoming changes affect your code, I can send you the complete driver. I'll post the changes to hdspm.c in a second, but that's just a preview for discussion, not necessarily the final patch. Cheers

2 3

Re: [alsa-devel] Please review rt5631 driver on alsa 1.0.24 again
by Mark Brown 23 Aug '11

23 Aug '11

On Mon, Aug 22, 2011 at 11:31:30AM +0800, johnnyhsu wrote: > +/* MIC Input Type */ > +static const char *rt5631_input_mode[] = { > + "Single-end", "Differential"}; Single ended. > +/* MIC Boost Volume */ > +static const char *rt5631_mic_boost[] = {"Bypass", "+20db", "+24db", "+30db", > + "+35db", "+40db", "+44db", "+50db", "+52db"}; TLV please. I'm fairly sure I mentioned this last time. > +/* Record Data Select */ > +static const char *rt5631_rec_sel[] = {"Disable", "MIC1", "MIC2", "Stereo"}; > + > +static const SOC_ENUM_SINGLE_DECL( > + rt5631_rec_sel_enum, RT5631_INT_ST_IRQ_CTRL_2, > + RT5631_ADC_DATA_SEL_SHIFT, rt5631_rec_sel); This looks like DAPM... > +/* SPK Channel Volume Input Select */ > +static const char *rt5631_spkvol_sel[] = {"VMID", "SPKMIX"}; > +static const SOC_ENUM_DOUBLE_DECL( > + rt5631_spkvol_enum, RT5631_SPK_OUT_VOL, > + RT5631_L_EN_SHIFT, RT5631_R_EN_SHIFT, rt5631_spkvol_sel); This too, and similarly for the other mixers. > +static void rt5631_set_dmic_params(struct snd_soc_codec *codec, > + struct snd_pcm_hw_params *params) > +{ > + int rate; > + > + snd_soc_update_bits(codec, RT5631_GPIO_CTRL, > + RT5631_GPIO_PIN_FUN_SEL_MASK | > + RT5631_GPIO_DMIC_FUN_SEL_MASK, > + RT5631_GPIO_PIN_FUN_SEL_GPIO_DIMC | > + RT5631_GPIO_DMIC_FUN_SEL_DIMC); Shouldn't this be done once in th eprobe function? DMICs aren't usually hotplugged... > + snd_soc_update_bits(codec, RT5631_DIG_MIC_CTRL, > + RT5631_DMIC_ENA_MASK, RT5631_DMIC_ENA); This looks like DAPM. > + snd_soc_update_bits(codec, RT5631_DIG_MIC_CTRL, > + RT5631_DMIC_L_CH_LATCH_MASK | > + RT5631_DMIC_R_CH_LATCH_MASK, > + RT5631_DMIC_L_CH_LATCH_FALLING | > + RT5631_DMIC_R_CH_LATCH_RISING); This also looks like probe work. > + default: > + break; > + } Should return an error if you don't know how to configure. > + snd_soc_update_bits(codec, RT5631_SDP_CTRL, > + RT5631_SDP_I2S_DL_MASK, iface); > + if (coeff >= 0) > + snd_soc_write(codec, RT5631_STEREO_AD_DA_CLK_CTRL, > + coeff_div[coeff].reg_val); What if you didn't find coefficients? > +static int rt5631_hifi_codec_set_dai_sysclk(struct snd_soc_dai *codec_dai, > + int clk_id, unsigned int freq, int dir) > +{ > + struct snd_soc_codec *codec = codec_dai->codec; > + struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec); > + > + pr_info("enter %s, syclk=%d\n", __func__, freq); > + if ((freq >= (256 * 8000)) && (freq <= (512 * 96000))) { > + rt5631->sysclk = freq; > + return 0; > + } > + > + pr_info("unsupported sysclk freq %u for audio i2s\n", freq); > + pr_info("set sysclk to 24.576Mhz by default\n"); > + > + rt5631->sysclk = 24576000; > + return 0; No, return an error if the user specified something you don't understand. If that's the sysclk the user can always specify it directly. > +static int rt5631_codec_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, > + int source, unsigned int freq_in, unsigned int freq_out) > +{ > + struct snd_soc_codec *codec = codec_dai->codec; > + struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec); > + int i, ret = -EINVAL; > + > + printk(KERN_DEBUG "enter %s\n", __func__); > + > + if (!freq_in || !freq_out) > + return 0; Should stop the PLL if a zero output is requested. > +/** > + * rt5631_index_reg_show - sysfs file for dumping index registers of 2nd layer > + */ > +#define IDX_REG_FMT "%02x: %04x\n" > +#define IDX_REG_LEN 9 > +static ssize_t rt5631_index_reg_show(struct device *dev, > + struct device_attribute *attr, char *buf) You didn't answer any of my queries about this stuff on the prior posting... > + ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C); > + if (ret != 0) { > + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); > + return ret; > + } > + codec->cache_bypass = 1; Why cache_bypass? > + /* speaker ratio is 1.44x */ > + snd_soc_write(codec, RT5631_GEN_PUR_CTRL_REG, 0x4e00); Hrm?

1 0

[alsa-devel] [PATCH 2/2] ASoC: mxs-sgtl5000: add record function
by Dong Aisheng 23 Aug '11

23 Aug '11

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> --- sound/soc/mxs/mxs-sgtl5000.c | 10 +++++++++- 1 files changed, 9 insertions(+), 1 deletions(-) diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index a0d89c9..7fbeaec 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -85,13 +85,21 @@ static struct snd_soc_ops mxs_sgtl5000_hifi_ops = { static struct snd_soc_dai_link mxs_sgtl5000_dai[] = { { - .name = "HiFi", + .name = "HiFi Tx", .stream_name = "HiFi Playback", .codec_dai_name = "sgtl5000", .codec_name = "sgtl5000.0-000a", .cpu_dai_name = "mxs-saif.0", .platform_name = "mxs-pcm-audio.0", .ops = &mxs_sgtl5000_hifi_ops, + }, { + .name = "HiFi Rx", + .stream_name = "HiFi Capture", + .codec_dai_name = "sgtl5000", + .codec_name = "sgtl5000.0-000a", + .cpu_dai_name = "mxs-saif.1", + .platform_name = "mxs-pcm-audio.1", + .ops = &mxs_sgtl5000_hifi_ops, }, }; -- 1.7.0.4

3 2

[alsa-devel] [PATCH 3/4] sound/soc/mxs/mxs-saif.c: add missing kfree
by Julia Lawall 23 Aug '11

23 Aug '11

From: Julia Lawall <julia(a)diku.dk> The error handling code prior to get_clk also needs to call kfree. The label name is changed to reflect its more general use. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ identifier x; expression E1!=0,E2,E3,E4; statement S; iterator I; @@ ( if (...) { ... when != kfree(x) when != x = E3 when != E3 = x * return ...; } ... when != x = E2 when != I(...,x,...) S if (...) { ... when != x = E4 kfree(x); ... return ...; } ) // </smpl> Signed-off-by: Julia Lawall <julia(a)diku.dk> --- sound/soc/mxs/mxs-saif.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 0b3adae..4a17bc8 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -523,8 +523,10 @@ static int mxs_saif_probe(struct platform_device *pdev) if (!saif) return -ENOMEM; - if (pdev->id >= ARRAY_SIZE(mxs_saif)) - return -EINVAL; + if (pdev->id >= ARRAY_SIZE(mxs_saif)) { + ret = -EINVAL; + goto failed; + } mxs_saif[pdev->id] = saif; saif->clk = clk_get(&pdev->dev, NULL); @@ -532,7 +534,7 @@ static int mxs_saif_probe(struct platform_device *pdev) ret = PTR_ERR(saif->clk); dev_err(&pdev->dev, "Cannot get the clock: %d\n", ret); - goto failed_clk; + goto failed; } res = platform_get_resource(pdev, IORESOURCE_MEM, 0); @@ -625,7 +627,7 @@ failed_ioremap: release_mem_region(res->start, resource_size(res)); failed_get_resource: clk_put(saif->clk); -failed_clk: +failed: kfree(saif); return ret;

5 9

[alsa-devel] [PATCH 1/1] ASoC: mxs-saif: clear clk gate first before register setting
by Dong Aisheng 23 Aug '11

23 Aug '11

Saif needs clear clk gate first before writing registers or the write will not success. The original xx_get_mclk function clear clk gate after mclk setting that may cause the former mclk setting unwork, then the real output mclk maybe inaccurate. Placing the clear before setting mclk to avoid such an issue. We also have to clear clk gate in startup instead of in prepare function. 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> --- sound/soc/mxs/mxs-saif.c | 22 ++++++++++++---------- 1 files changed, 12 insertions(+), 10 deletions(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 0b3adae..530017f 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -187,16 +187,20 @@ int mxs_saif_get_mclk(unsigned int saif_id, unsigned int mclk, if (!saif) return -EINVAL; + /* Clear Reset */ + __raw_writel(BM_SAIF_CTRL_SFTRST, + saif->base + SAIF_CTRL + MXS_CLR_ADDR); + + /* FIXME: need clear clk gate for register r/w */ + __raw_writel(BM_SAIF_CTRL_CLKGATE, + saif->base + SAIF_CTRL + MXS_CLR_ADDR); + stat = __raw_readl(saif->base + SAIF_STAT); if (stat & BM_SAIF_STAT_BUSY) { dev_err(saif->dev, "error: busy\n"); return -EBUSY; } - /* Clear Reset */ - __raw_writel(BM_SAIF_CTRL_SFTRST, - saif->base + SAIF_CTRL + MXS_CLR_ADDR); - saif->mclk_in_use = 1; ret = mxs_saif_set_clk(saif, mclk, rate); if (ret) @@ -207,8 +211,6 @@ int mxs_saif_get_mclk(unsigned int saif_id, unsigned int mclk, return ret; /* enable MCLK output */ - __raw_writel(BM_SAIF_CTRL_CLKGATE, - saif->base + SAIF_CTRL + MXS_CLR_ADDR); __raw_writel(BM_SAIF_CTRL_RUN, saif->base + SAIF_CTRL + MXS_SET_ADDR); @@ -303,6 +305,10 @@ static int mxs_saif_startup(struct snd_pcm_substream *substream, __raw_writel(BM_SAIF_CTRL_SFTRST, saif->base + SAIF_CTRL + MXS_CLR_ADDR); + /* clear clock gate */ + __raw_writel(BM_SAIF_CTRL_CLKGATE, + saif->base + SAIF_CTRL + MXS_CLR_ADDR); + return 0; } @@ -379,10 +385,6 @@ static int mxs_saif_prepare(struct snd_pcm_substream *substream, { struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai); - /* clear clock gate */ - __raw_writel(BM_SAIF_CTRL_CLKGATE, - saif->base + SAIF_CTRL + MXS_CLR_ADDR); - /* enable FIFO error irqs */ __raw_writel(BM_SAIF_CTRL_FIFO_ERROR_IRQ_EN, saif->base + SAIF_CTRL + MXS_SET_ADDR); -- 1.7.0.4

3 2

[alsa-devel] [PATCH] [v2] sound/soc/fsl/fsl_dma.c: add missing of_node_put
by Timur Tabi 23 Aug '11

23 Aug '11

of_parse_phandle increments the reference count of np, so this should be decremented before trying the next possibility. Since we don't actually use np, we can decrement the reference count immediately. Reported-by: Julia Lawall <julia(a)diku.dk> Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/fsl_dma.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 6680c0b..b300f4b 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -877,10 +877,12 @@ static struct device_node *find_ssi_node(struct device_node *dma_channel_np) * assume that device_node pointers are a valid comparison. */ np = of_parse_phandle(ssi_np, "fsl,playback-dma", 0); + of_node_put(np); if (np == dma_channel_np) return ssi_np; np = of_parse_phandle(ssi_np, "fsl,capture-dma", 0); + of_node_put(np); if (np == dma_channel_np) return ssi_np; } -- 1.7.3.4

3 3

[alsa-devel] [PATCH 2/2] sound/soc/fsl/mpc8610_hpcd.c: add missing of_node_put
by Julia Lawall 23 Aug '11

23 Aug '11

From: Julia Lawall <julia(a)diku.dk> The first change is to add an of_node_put, since codec_np has previously been allocated. The rest of the patch reorganizes the error handling code so the only code executed is that which is needed. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ identifier x; expression E1!=0,E2,E3,E4; statement S; iterator I; @@ ( if (...) { ... when != of_node_put(x) when != x = E3 when != E3 = x * return ...; } ... when != x = E2 when != I(...,x,...) S if (...) { ... when != x = E4 of_node_put(x); ... return ...; } ) // </smpl> Signed-off-by: Julia Lawall <julia(a)diku.dk> --- sound/soc/fsl/mpc8610_hpcd.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index a192979..358f0ba 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -345,8 +345,10 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) } machine_data = kzalloc(sizeof(struct mpc8610_hpcd_data), GFP_KERNEL); - if (!machine_data) - return -ENOMEM; + if (!machine_data) { + ret = -ENOMEM; + goto error_alloc; + } machine_data->dai[0].cpu_dai_name = dev_name(&ssi_pdev->dev); machine_data->dai[0].ops = &mpc8610_hpcd_ops; @@ -494,7 +496,7 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) ret = platform_device_add(sound_device); if (ret) { dev_err(&pdev->dev, "platform device add failed\n"); - goto error; + goto error_sound; } dev_set_drvdata(&pdev->dev, sound_device); @@ -502,14 +504,12 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) return 0; +error_sound: + platform_device_unregister(sound_device); error: - of_node_put(codec_np); - - if (sound_device) - platform_device_unregister(sound_device); - kfree(machine_data); - +error_alloc: + of_node_put(codec_np); return ret; }

2 1

[alsa-devel] [PATCH 1/2] sound/soc/fsl/p1022_ds.c: add missing of_node_put
by Julia Lawall 23 Aug '11

23 Aug '11

From: Julia Lawall <julia(a)diku.dk> dma_channel_np has been accessed at this point, so decrease its reference count before leaving the function. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ identifier x; expression E1!=0,E2,E3,E4; statement S; iterator I; @@ ( if (...) { ... when != of_node_put(x) when != x = E3 when != E3 = x * return ...; } ... when != x = E2 when != I(...,x,...) S if (...) { ... when != x = E4 of_node_put(x); ... return ...; } ) // </smpl> Signed-off-by: Julia Lawall <julia(a)diku.dk> --- sound/soc/fsl/p1022_ds.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index 8fa4d5f..fcb862e 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -297,8 +297,10 @@ static int get_dma_channel(struct device_node *ssi_np, * dai->platform name should already point to an allocated buffer. */ ret = of_address_to_resource(dma_channel_np, 0, &res); - if (ret) + if (ret) { + of_node_put(dma_channel_np); return ret; + } snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s", (unsigned long long) res.start, dma_channel_np->name);

4 3

[alsa-devel] [PATCH 2/4] sound/soc/ep93xx/ep93xx-i2s.c: add missing kfree
by Julia Lawall 23 Aug '11

23 Aug '11

From: Julia Lawall <julia(a)diku.dk> Introduce a new label that includes kfree and jump to that one. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ identifier x; expression E1!=0,E2,E3,E4; statement S; iterator I; @@ ( if (...) { ... when != kfree(x) when != x = E3 when != E3 = x * return ...; } ... when != x = E2 when != I(...,x,...) S if (...) { ... when != x = E4 kfree(x); ... return ...; } ) // </smpl> Signed-off-by: Julia Lawall <julia(a)diku.dk> --- sound/soc/ep93xx/ep93xx-i2s.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c index 56efa0c..099614e 100644 --- a/sound/soc/ep93xx/ep93xx-i2s.c +++ b/sound/soc/ep93xx/ep93xx-i2s.c @@ -385,14 +385,14 @@ static int ep93xx_i2s_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!res) { err = -ENODEV; - goto fail; + goto fail_free_info; } info->mem = request_mem_region(res->start, resource_size(res), pdev->name); if (!info->mem) { err = -EBUSY; - goto fail; + goto fail_free_info; } info->regs = ioremap(info->mem->start, resource_size(info->mem)); @@ -435,6 +435,7 @@ fail_unmap_mem: iounmap(info->regs); fail_release_mem: release_mem_region(info->mem->start, resource_size(info->mem)); +fail_free_info: kfree(info); fail: return err;

3 2

Re: [alsa-devel] [PATCH] ASoC: soc-core: use GFP_KERNEL flag for kmalloc in snd_soc_cnew
by Mark Brown 23 Aug '11

23 Aug '11

On Sat, Aug 20, 2011 at 11:03:44AM +0800, Axel Lin wrote: > GFP_ATOMIC is not needed here, use GFP_KERNEL instead. > > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> Applied, thanks.

1 0

[alsa-devel] [PATCH] ASoC: Allow idle_bias_off to be specified in CODEC drivers
by Mark Brown 22 Aug '11

22 Aug '11

If devices can unconditionally support idle_bias_off let them flag it in their driver structure. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 1 + sound/soc/soc-core.c | 2 ++ 2 files changed, 3 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 3fe658e..6da55a1 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -633,6 +633,7 @@ struct snd_soc_codec_driver { /* codec bias level */ int (*set_bias_level)(struct snd_soc_codec *, enum snd_soc_bias_level level); + bool idle_bias_off; void (*seq_notifier)(struct snd_soc_dapm_context *, enum snd_soc_dapm_type, int); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ae93aa8..f8f985a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -956,6 +956,8 @@ static int soc_probe_codec(struct snd_soc_card *card, snd_soc_dapm_new_controls(&codec->dapm, driver->dapm_widgets, driver->num_dapm_widgets); + codec->dapm.idle_bias_off = driver->idle_bias_off; + if (driver->probe) { ret = driver->probe(codec); if (ret < 0) { -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: Convert WM8523 to table based control and DAPM initialization
by Mark Brown 22 Aug '11

22 Aug '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8523.c | 26 +++++++++----------------- 1 files changed, 9 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 52812d1..5355a7a 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -85,7 +85,7 @@ static const char *wm8523_zd_count_text[] = { static const struct soc_enum wm8523_zc_count = SOC_ENUM_SINGLE(WM8523_ZERO_DETECT, 0, 2, wm8523_zd_count_text); -static const struct snd_kcontrol_new wm8523_snd_controls[] = { +static const struct snd_kcontrol_new wm8523_controls[] = { SOC_DOUBLE_R_TLV("Playback Volume", WM8523_DAC_GAINL, WM8523_DAC_GAINR, 0, 448, 0, dac_tlv), SOC_SINGLE("ZC Switch", WM8523_DAC_CTRL3, 4, 1, 0), @@ -102,22 +102,11 @@ SND_SOC_DAPM_OUTPUT("LINEVOUTL"), SND_SOC_DAPM_OUTPUT("LINEVOUTR"), }; -static const struct snd_soc_dapm_route intercon[] = { +static const struct snd_soc_dapm_route wm8523_dapm_routes[] = { { "LINEVOUTL", NULL, "DAC" }, { "LINEVOUTR", NULL, "DAC" }, }; -static int wm8523_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, wm8523_dapm_widgets, - ARRAY_SIZE(wm8523_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); - - return 0; -} - static struct { int value; int ratio; @@ -480,10 +469,6 @@ static int wm8523_probe(struct snd_soc_codec *codec) /* Bias level configuration will have done an extra enable */ regulator_bulk_disable(ARRAY_SIZE(wm8523->supplies), wm8523->supplies); - snd_soc_add_controls(codec, wm8523_snd_controls, - ARRAY_SIZE(wm8523_snd_controls)); - wm8523_add_widgets(codec); - return 0; err_enable: @@ -513,6 +498,13 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8523 = { .reg_word_size = sizeof(u16), .reg_cache_default = wm8523_reg, .volatile_register = wm8523_volatile_register, + + .controls = wm8523_controls, + .num_controls = ARRAY_SIZE(wm8523_controls), + .dapm_widgets = wm8523_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8523_dapm_widgets), + .dapm_routes = wm8523_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8523_dapm_routes), }; static const struct of_device_id wm8523_of_match[] = { -- 1.7.5.4

2 1

[alsa-devel] about UCM on bluetooth headset
by Lu Guanqun 22 Aug '11

22 Aug '11

Hi Liam, In alsa-libs:include/use-case.h, there are below lines: * a conversation etc. The device describes the physical audio capture and playback * hardware i.e. headphones, phone handset, bluetooth headset, etc. I'm not quite clear about how to set a use case for bluetooth headset. According to my understanding, bluetooth doesn't fall into the category of ALSA, therefore there's no such direct devices for it. Could you shed some lights on this? What's the best practises to use UCM with bluetooth? -- guanqun

3 3

[alsa-devel] [PATCH] sound/soc/fsl/fsl_dma.c: add missing of_node_put
by Julia Lawall 22 Aug '11

22 Aug '11

From: Julia Lawall <julia(a)diku.dk> of_parse_phandle increments the reference count of np, so this should be decremented before trying the next possibility. The semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ expression e,e1,e2; @@ *e = of_parse_phandle(...) ... when != of_node_put(e) when != true e == NULL when != e2 = e e = e1 // </smpl> Signed-off-by: Julia Lawall <julia(a)diku.dk> --- sound/soc/fsl/fsl_dma.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 0efc04a..b33271b 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -880,10 +880,12 @@ static struct device_node *find_ssi_node(struct device_node *dma_channel_np) np = of_parse_phandle(ssi_np, "fsl,playback-dma", 0); if (np == dma_channel_np) return ssi_np; + of_node_put(np); np = of_parse_phandle(ssi_np, "fsl,capture-dma", 0); if (np == dma_channel_np) return ssi_np; + of_node_put(np); } return NULL;

3 3

Re: [alsa-devel] [PATCH 2/2] USB: Gadget: Add Audio Class 2.0 Driver
by Jassi Brar 22 Aug '11

22 Aug '11

On Mon, Aug 22, 2011 at 7:16 PM, Felipe Balbi <balbi(a)ti.com> wrote: > >> But that would mean having two drivers in one file, because they are >> so different. > > then make only a differnt function driver, f_uac2.c, which gets added as > another interface (or configuration) to audio.c. Yup, by audio.c I meant {audio.c + f_audio.c} and audio2.c {audio2.c + f_audio2.c} >> Otherwise, please have a look at audio.c and audio2.c and suggest what >> exactly you mean. > > audio.c doesn't do anything audio related. It's just adding the > configuration to composite layer. The real specification is implemented > in f_audio.c (should probably be called f_uac1.c, though). Agreed. Reusing audio.c is quite reasonable. So, I'll :- * Merge audio2.c into audio.c * Rename f_audio.c -> f_uac1.c * Add new f_audio2.c as f_uac2.c * Add Kconfig switch to compile UAC_1 (audio.c + f_uac1.c) or UAC_2 (audio.c + f_uac2.c) I'll wait for other comments and revise the patch after a few days. Thanks.

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Re: [alsa-devel] [PATCH 2/2] USB: Gadget: Add Audio Class 2.0 Driver
by Jassi Brar 22 Aug '11

22 Aug '11

On Mon, Aug 22, 2011 at 6:14 PM, Felipe Balbi <balbi(a)ti.com> wrote: >> Functionality wise, audio.c hardcodes the data-path to the physical >> audio-codec onboard >> whereas audio2.c provides user-space with a virtual sound-card to fool around >> with audio-data the way the user wants. The functionality that has >> lion's share of code. >> Also, although the functionality of audio.c can be achieved by audio2.c (but not >> vice-versa) we'll break user-space by changing the behavior of audio.c >> to audio2.c >> >> Besides fundamental differences in functionality, they differ by >> Usb-Audio-Class >> compliance as well. audio.c is UAC_1 compliant, whereas audio2.c is UAC_2 >> complaint. >> One can not replace the other. For ex, MS-Windows doesn't support >> UAC_2 by default >> yet, Linux does reasonably well. And by definition having >> Linux-USB-Gadget support >> UAC_2 also, is only better. So by having both we cover more ground than by >> having just one. > > I understand your point, but then we will have two gadget drivers (and > two function drivers) for the same thing (USB Audio), it's the same > thing with the storage gadgets and we have decided on dropping one of > them recently, so I'm not keen on accepting another gadget driver which > will essentially do the same thing. No dear, not the same thing. As I said, they differ by features as well as USB-Spec... only the 'audio' thing is common. > If audio2.c can achieve audio.c's functionality, then you can start by > re-factoring audio.c so that it come closer to what audio2.c is today. audio.c and audio2.c have almost nothing in common. Please have a look at the code. Making audio.c closer to audio2.c would mean :- * Changing all of the descriptors of audio.c {uac_1 -> uac_2} * Removing all alsa related code and adding virtual alsa sound card. At the end, audio.c would have been changed completely and Linux-usb-gadget lost the UAC_1 compliance. > WRT UAC_1/UAC_2 compliance, you can use a Kconfig entry, or two separate > USB configurations (one with UAC_1 and another UAC_2). So there are ways > to make this all work while still keeping backwards compatibility. IIUC, you mean adding UAC_2 compliance and virtual sound-card i/f to audio.c, while preserving the it's original functionality ? But that would mean having two drivers in one file, because they are so different. Otherwise, please have a look at audio.c and audio2.c and suggest what exactly you mean. thnx

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Re: [alsa-devel] [PATCH] ASoC: soc-core: use GFP_KERNEL flag for kmalloc in snd_soc_cnew
by Liam Girdwood 22 Aug '11

22 Aug '11

On 20/08/11 04:03, Axel Lin wrote: > GFP_ATOMIC is not needed here, use GFP_KERNEL instead. > > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> > --- > sound/soc/soc-core.c | 2 +- > 1 files changed, 1 insertions(+), 1 deletions(-) > > diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c > index ae93aa8..b6d8716 100644 > --- a/sound/soc/soc-core.c > +++ b/sound/soc/soc-core.c > @@ -1913,7 +1913,7 @@ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, > > if (prefix) { > name_len = strlen(long_name) + strlen(prefix) + 2; > - name = kmalloc(name_len, GFP_ATOMIC); > + name = kmalloc(name_len, GFP_KERNEL); > if (!name) > return NULL; > Acked-by: Liam Girdwood <lrg(a)ti.com>

1 0

Re: [alsa-devel] [PATCH] ASoC: Provide more detail on WM8962 thermal shutdown status
by Liam Girdwood 22 Aug '11

22 Aug '11

+ alsa-devel On 21/08/11 18:08, Mark Brown wrote: > Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> > --- Acked-by: Liam Girdwood <lrg(a)ti.com> > sound/soc/codecs/wm8962.c | 17 +++++++++++++++-- > 1 files changed, 15 insertions(+), 2 deletions(-) > > diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c > index d2db512..ae7a263 100644 > --- a/sound/soc/codecs/wm8962.c > +++ b/sound/soc/codecs/wm8962.c > @@ -839,7 +839,7 @@ static const struct wm8962_reg_access { > [40] = { 0x00FF, 0x01FF, 0x0000 }, /* R40 - SPKOUTL volume */ > [41] = { 0x00FF, 0x01FF, 0x0000 }, /* R41 - SPKOUTR volume */ > > - [47] = { 0x000F, 0x0000, 0x0000 }, /* R47 - Thermal Shutdown Status */ > + [47] = { 0x000F, 0x0000, 0xFFFF }, /* R47 - Thermal Shutdown Status */ > [48] = { 0x7EC7, 0x7E07, 0xFFFF }, /* R48 - Additional Control (4) */ > [49] = { 0x00D3, 0x00D7, 0xFFFF }, /* R49 - Class D Control 1 */ > [51] = { 0x0047, 0x0047, 0x0000 }, /* R51 - Class D Control 2 */ > @@ -3575,6 +3575,7 @@ static irqreturn_t wm8962_irq(int irq, void *data) > struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); > int mask; > int active; > + int reg; > > mask = snd_soc_read(codec, WM8962_INTERRUPT_STATUS_2_MASK); > > @@ -3595,9 +3596,21 @@ static irqreturn_t wm8962_irq(int irq, void *data) > if (active & WM8962_FIFOS_ERR_EINT) > dev_err(codec->dev, "FIFO error\n"); > > - if (active & WM8962_TEMP_SHUT_EINT) > + if (active & WM8962_TEMP_SHUT_EINT) { > dev_crit(codec->dev, "Thermal shutdown\n"); > > + reg = snd_soc_read(codec, WM8962_THERMAL_SHUTDOWN_STATUS); > + > + if (reg & WM8962_TEMP_ERR_HP) > + dev_crit(codec->dev, "Headphone thermal error\n"); > + if (reg & WM8962_TEMP_WARN_HP) > + dev_crit(codec->dev, "Headphone thermal warning\n"); > + if (reg & WM8962_TEMP_ERR_SPK) > + dev_crit(codec->dev, "Speaker thermal error\n"); > + if (reg & WM8962_TEMP_WARN_SPK) > + dev_crit(codec->dev, "Speaker thermal warning\n"); > + } > + > if (active & (WM8962_MICSCD_EINT | WM8962_MICD_EINT)) { > dev_dbg(codec->dev, "Microphone event detected\n"); >

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Re: [alsa-devel] asoc: no valid backend routes for PCM
by Liam Girdwood 22 Aug '11

22 Aug '11

Please don't drop the list from replies. On 22/08/11 06:38, sarika baby wrote: > Hello, > Thanks for your information, but can you tell that if i have to add the audio route should i have to do it at kernel drivers level or else at ALSA (mydroid level) ? > The routes are all configured in userspace with either UCM, alsamixer, amixer or by calling the alsa-lib C API. Liam > Thanks and Regards > Soujanya > > On Fri, Aug 19, 2011 at 9:50 PM, Liam Girdwood <lrg(a)ti.com <mailto:lrg@ti.com>> wrote: > > On 19/08/11 08:53, Sarika wrote: > > Hello, > > I am not able to hear audio on my headset when it connect it to panda board. > > What mite be the issue? > > This means you have no valid audio route from your pcm device to your headphone. > > Please use the supplied UCM configuration files (with Ubuntu) to set the route or use alsamixer to do this manually. > > http://www.omappedia.org/wiki/Audio_Drive_Arch > > > Also panda does not has DMIC does this make any difference ? > > > > No. > > Liam > > > Regards > > Sarika > > > > > > _______________________________________________ > > Alsa-devel mailing list > > Alsa-devel(a)alsa-project.org <mailto:Alsa-devel@alsa-project.org> > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > > > > -- > SarikA >

1 0

[alsa-devel] [PATCH] ucm: change msleep to usleep
by Lu Guanqun 22 Aug '11

22 Aug '11

In file src/ucm/parser.c: if (strcmp(cmd, "usleep") == 0) { string `usleep' is compared, however, in the comment and example conf file, `msleep' is used, it's better to unify them all. Signed-off-by: Lu Guanqun <guanqun.lu(a)intel.com> --- src/ucm/parser.c | 8 ++++---- test/ucm/TestHDA/TestHDA.conf | 2 +- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/src/ucm/parser.c b/src/ucm/parser.c index 23b67bc..a56f1e6 100644 --- a/src/ucm/parser.c +++ b/src/ucm/parser.c @@ -792,7 +792,7 @@ static int parse_modifier_name(snd_use_case_mgr_t *uc_mgr, * EnableSequence [ * cset "name='Master Playback Switch',index=2 0,0" * cset "name='Master Playback Volume',index=2 25,25" - * msleep 50 + * usleep 50 * cset "name='Master Playback Switch',index=2 1,1" * cset "name='Master Playback Volume',index=2 50,50" * ] @@ -800,14 +800,14 @@ static int parse_modifier_name(snd_use_case_mgr_t *uc_mgr, * DisableSequence [ * cset "name='Master Playback Switch',index=2 0,0" * cset "name='Master Playback Volume',index=2 25,25" - * msleep 50 + * usleep 50 * cset "name='Master Playback Switch',index=2 1,1" * cset "name='Master Playback Volume',index=2 50,50" * ] * * # Optional transition verb * TransitionSequence."ToCaseName" [ - * msleep 1 + * usleep 1 * ] * * # Optional TQ and ALSA PCMs @@ -1106,7 +1106,7 @@ static int parse_controls(snd_use_case_mgr_t *uc_mgr, snd_config_t *cfg) * cset "name='Master Mono Playback Volume',index=1 0" * cset "name='PCM Switch',index=2 1,1" * exec "some binary here" - * msleep 50 + * usleep 50 * ........ * ] * diff --git a/test/ucm/TestHDA/TestHDA.conf b/test/ucm/TestHDA/TestHDA.conf index 41dd74c..0a75e21 100644 --- a/test/ucm/TestHDA/TestHDA.conf +++ b/test/ucm/TestHDA/TestHDA.conf @@ -7,7 +7,7 @@ SectionUseCase."Case1" { SectionDefaults [ exec "my prg" - msleep 1 + usleep 1 cdev "hw:0" cset "name='PCM Playback Volume' 50%" ]

3 8

[alsa-devel] [PATCH] ASoC: Clear any outstanding WM8962 FLL lock completions before waiting
by Mark Brown 22 Aug '11

22 Aug '11

Ensure that we don't spuriously trigger early. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 4 ++++ 1 files changed, 4 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index c9e07aa..d2db512 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2356,6 +2356,8 @@ static int sysclk_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: if (fll) { + try_wait_for_completion(&wm8962->fll_lock); + snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1, WM8962_FLL_ENA, WM8962_FLL_ENA); if (wm8962->irq) { @@ -3456,6 +3458,8 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source, snd_soc_write(codec, WM8962_FLL_CONTROL_7, fll_div.lambda); snd_soc_write(codec, WM8962_FLL_CONTROL_8, fll_div.n); + try_wait_for_completion(&wm8962->fll_lock); + snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1, WM8962_FLL_FRAC | WM8962_FLL_REFCLK_SRC_MASK | WM8962_FLL_ENA, fll1); -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: Ensure we get an impedence reported for WM8958 jack detect
by Mark Brown 22 Aug '11

22 Aug '11

Occasionally we may see an accessory reported before we have a stable impedance for the accessory. If this happens then reread the status in order to ensure that the handler can take the appropriate action for the status change. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/linux/mfd/wm8994/registers.h | 15 +++++++++++++ sound/soc/codecs/wm8994.c | 37 +++++++++++++++++++++++---------- 2 files changed, 41 insertions(+), 11 deletions(-) diff --git a/include/linux/mfd/wm8994/registers.h b/include/linux/mfd/wm8994/registers.h index d7be463..10c483f 100644 --- a/include/linux/mfd/wm8994/registers.h +++ b/include/linux/mfd/wm8994/registers.h @@ -1967,6 +1967,21 @@ #define WM8958_MICB2_DISCH_WIDTH 1 /* MICB2_DISCH */ /* + * R210 (0xD2) - Mic Detect 3 + */ +#define WM8958_MICD_LVL_MASK 0x07FC /* MICD_LVL - [10:2] */ +#define WM8958_MICD_LVL_SHIFT 2 /* MICD_LVL - [10:2] */ +#define WM8958_MICD_LVL_WIDTH 9 /* MICD_LVL - [10:2] */ +#define WM8958_MICD_VALID 0x0002 /* MICD_VALID */ +#define WM8958_MICD_VALID_MASK 0x0002 /* MICD_VALID */ +#define WM8958_MICD_VALID_SHIFT 1 /* MICD_VALID */ +#define WM8958_MICD_VALID_WIDTH 1 /* MICD_VALID */ +#define WM8958_MICD_STS 0x0001 /* MICD_STS */ +#define WM8958_MICD_STS_MASK 0x0001 /* MICD_STS */ +#define WM8958_MICD_STS_SHIFT 0 /* MICD_STS */ +#define WM8958_MICD_STS_WIDTH 1 /* MICD_STS */ + +/* * R76 (0x4C) - Charge Pump (1) */ #define WM8994_CP_ENA 0x8000 /* CP_ENA */ diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 5e8d66d..a26cee1 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3043,19 +3043,34 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) { struct wm8994_priv *wm8994 = data; struct snd_soc_codec *codec = wm8994->codec; - int reg; + int reg, count; - reg = snd_soc_read(codec, WM8958_MIC_DETECT_3); - if (reg < 0) { - dev_err(codec->dev, "Failed to read mic detect status: %d\n", - reg); - return IRQ_NONE; - } + /* We may occasionally read a detection without an impedence + * range being provided - if that happens loop again. + */ + count = 10; + do { + reg = snd_soc_read(codec, WM8958_MIC_DETECT_3); + if (reg < 0) { + dev_err(codec->dev, + "Failed to read mic detect status: %d\n", + reg); + return IRQ_NONE; + } - if (!(reg & WM8958_MICD_VALID)) { - dev_dbg(codec->dev, "Mic detect data not valid\n"); - goto out; - } + if (!(reg & WM8958_MICD_VALID)) { + dev_dbg(codec->dev, "Mic detect data not valid\n"); + goto out; + } + + if (!(reg & WM8958_MICD_STS) || (reg & WM8958_MICD_LVL_MASK)) + break; + + msleep(1); + } while (count--); + + if (count == 0) + dev_warn(codec->dev, "No impedence range reported for jack\n"); #ifndef CONFIG_SND_SOC_WM8994_MODULE trace_snd_soc_jack_irq(dev_name(codec->dev)); -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: Report IRQ_NONE when we don't see an interrupt from WM8962
by Mark Brown 22 Aug '11

22 Aug '11

This should never happen with level triggered IRQs. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 0c61c2f..b877fa9 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3577,6 +3577,9 @@ static irqreturn_t wm8962_irq(int irq, void *data) active = snd_soc_read(codec, WM8962_INTERRUPT_STATUS_2); active &= ~mask; + if (!active) + return IRQ_NONE; + /* Acknowledge the interrupts */ snd_soc_write(codec, WM8962_INTERRUPT_STATUS_2, active); -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: Initial WM8962 DSP2 support
by Mark Brown 22 Aug '11

22 Aug '11

The WM8962 features a DSP providing a number of signal processing features including HD Bass and Virtual Surround Sound (VSS). Enable initial support for this, allowing users to enable and disable the algorithms using the default coefficient sets. Further patches will add support for runtime configuration of the DSP coefficients. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 157 ++++++++++++++++++++++++++++++++++++++++++++- 1 files changed, 156 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 8526383..c9e07aa 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -63,6 +63,8 @@ struct wm8962_priv { int fll_fref; int fll_fout; + u16 dsp2_ena; + struct delayed_work mic_work; struct snd_soc_jack *jack; @@ -967,7 +969,7 @@ static const struct wm8962_reg_access { [584] = { 0x002D, 0x002D, 0x0000 }, /* R584 - IRQ Debounce */ [586] = { 0xC000, 0xC000, 0x0000 }, /* R586 - MICINT Source Pol */ [768] = { 0x0001, 0x0001, 0x0000 }, /* R768 - DSP2 Power Management */ - [1037] = { 0x0000, 0x003F, 0x0000 }, /* R1037 - DSP2_ExecControl */ + [1037] = { 0x0000, 0x003F, 0xFFFF }, /* R1037 - DSP2_ExecControl */ [4096] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4096 - Write Sequencer 0 */ [4097] = { 0x00FF, 0x00FF, 0x0000 }, /* R4097 - Write Sequencer 1 */ [4098] = { 0x070F, 0x070F, 0x0000 }, /* R4098 - Write Sequencer 2 */ @@ -1988,6 +1990,122 @@ static const unsigned int classd_tlv[] = { }; static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); +static int wm8962_dsp2_write_config(struct snd_soc_codec *codec) +{ + return 0; +} + +static int wm8962_dsp2_set_enable(struct snd_soc_codec *codec, u16 val) +{ + u16 adcl = snd_soc_read(codec, WM8962_LEFT_ADC_VOLUME); + u16 adcr = snd_soc_read(codec, WM8962_RIGHT_ADC_VOLUME); + u16 dac = snd_soc_read(codec, WM8962_ADC_DAC_CONTROL_1); + + /* Mute the ADCs and DACs */ + snd_soc_write(codec, WM8962_LEFT_ADC_VOLUME, 0); + snd_soc_write(codec, WM8962_RIGHT_ADC_VOLUME, WM8962_ADC_VU); + snd_soc_update_bits(codec, WM8962_ADC_DAC_CONTROL_1, + WM8962_DAC_MUTE, WM8962_DAC_MUTE); + + snd_soc_write(codec, WM8962_SOUNDSTAGE_ENABLES_0, val); + + /* Restore the ADCs and DACs */ + snd_soc_write(codec, WM8962_LEFT_ADC_VOLUME, adcl); + snd_soc_write(codec, WM8962_RIGHT_ADC_VOLUME, adcr); + snd_soc_update_bits(codec, WM8962_ADC_DAC_CONTROL_1, + WM8962_DAC_MUTE, dac); + + return 0; +} + +static int wm8962_dsp2_start(struct snd_soc_codec *codec) +{ + struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); + + wm8962_dsp2_write_config(codec); + + snd_soc_write(codec, WM8962_DSP2_EXECCONTROL, WM8962_DSP2_RUNR); + + wm8962_dsp2_set_enable(codec, wm8962->dsp2_ena); + + return 0; +} + +static int wm8962_dsp2_stop(struct snd_soc_codec *codec) +{ + wm8962_dsp2_set_enable(codec, 0); + + snd_soc_write(codec, WM8962_DSP2_EXECCONTROL, WM8962_DSP2_STOP); + + return 0; +} + +#define WM8962_DSP2_ENABLE(xname, xshift) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = wm8962_dsp2_ena_info, \ + .get = wm8962_dsp2_ena_get, .put = wm8962_dsp2_ena_put, \ + .private_value = xshift } + +static int wm8962_dsp2_ena_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + + return 0; +} + +static int wm8962_dsp2_ena_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int shift = kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = !!(wm8962->dsp2_ena & 1 << shift); + + return 0; +} + +static int wm8962_dsp2_ena_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int shift = kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); + int old = wm8962->dsp2_ena; + int ret = 0; + int dsp2_running = snd_soc_read(codec, WM8962_DSP2_POWER_MANAGEMENT) & + WM8962_DSP2_ENA; + + mutex_lock(&codec->mutex); + + if (ucontrol->value.integer.value[0]) + wm8962->dsp2_ena |= 1 << shift; + else + wm8962->dsp2_ena &= ~(1 << shift); + + if (wm8962->dsp2_ena == old) + goto out; + + ret = 1; + + if (dsp2_running) { + if (wm8962->dsp2_ena) + wm8962_dsp2_set_enable(codec, wm8962->dsp2_ena); + else + wm8962_dsp2_stop(codec); + } + +out: + mutex_unlock(&codec->mutex); + + return ret; +} + /* The VU bits for the headphones are in a different register to the mute * bits and only take effect on the PGA if it is actually powered. */ @@ -2146,6 +2264,11 @@ SOC_DOUBLE_R_TLV("EQ4 Volume", WM8962_EQ3, WM8962_EQ23, WM8962_EQL_B4_GAIN_SHIFT, 31, 0, eq_tlv), SOC_DOUBLE_R_TLV("EQ5 Volume", WM8962_EQ3, WM8962_EQ23, WM8962_EQL_B5_GAIN_SHIFT, 31, 0, eq_tlv), + +WM8962_DSP2_ENABLE("VSS Switch", WM8962_VSS_ENA_SHIFT), +WM8962_DSP2_ENABLE("HPF1 Switch", WM8962_HPF1_ENA_SHIFT), +WM8962_DSP2_ENABLE("HPF2 Switch", WM8962_HPF2_ENA_SHIFT), +WM8962_DSP2_ENABLE("HD Bass Switch", WM8962_HDBASS_ENA_SHIFT), }; static const struct snd_kcontrol_new wm8962_spk_mono_controls[] = { @@ -2405,6 +2528,31 @@ static int out_pga_event(struct snd_soc_dapm_widget *w, } } +static int dsp2_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (wm8962->dsp2_ena) + wm8962_dsp2_start(codec); + break; + + case SND_SOC_DAPM_PRE_PMD: + if (wm8962->dsp2_ena) + wm8962_dsp2_stop(codec); + break; + + default: + BUG(); + return -EINVAL; + } + + return 0; +} + static const char *st_text[] = { "None", "Right", "Left" }; static const struct soc_enum str_enum = @@ -2527,6 +2675,9 @@ SND_SOC_DAPM_SUPPLY("SYSCLK", WM8962_CLOCKING2, 5, 0, sysclk_event, SND_SOC_DAPM_SUPPLY("Charge Pump", WM8962_CHARGE_PUMP_1, 0, 0, cp_event, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("TOCLK", WM8962_ADDITIONAL_CONTROL_1, 0, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY_S("DSP2", 1, WM8962_DSP2_POWER_MANAGEMENT, + WM8962_DSP2_ENA_SHIFT, 0, dsp2_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_MIXER("INPGAL", WM8962_LEFT_INPUT_PGA_CONTROL, 4, 0, inpgal, ARRAY_SIZE(inpgal)), @@ -2622,11 +2773,13 @@ static const struct snd_soc_dapm_route wm8962_intercon[] = { { "ADCL", NULL, "TOCLK" }, { "ADCL", NULL, "MIXINL" }, { "ADCL", NULL, "DMIC" }, + { "ADCL", NULL, "DSP2" }, { "ADCR", NULL, "SYSCLK" }, { "ADCR", NULL, "TOCLK" }, { "ADCR", NULL, "MIXINR" }, { "ADCR", NULL, "DMIC" }, + { "ADCR", NULL, "DSP2" }, { "STL", "Left", "ADCL" }, { "STL", "Right", "ADCR" }, @@ -2638,11 +2791,13 @@ static const struct snd_soc_dapm_route wm8962_intercon[] = { { "DACL", NULL, "TOCLK" }, { "DACL", NULL, "Beep" }, { "DACL", NULL, "STL" }, + { "DACL", NULL, "DSP2" }, { "DACR", NULL, "SYSCLK" }, { "DACR", NULL, "TOCLK" }, { "DACR", NULL, "Beep" }, { "DACR", NULL, "STR" }, + { "DACR", NULL, "DSP2" }, { "HPMIXL", "IN4L Switch", "IN4L" }, { "HPMIXL", "IN4R Switch", "IN4R" }, -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: Ensure we only run Speyside WM8962 bias level callbacks once
by Mark Brown 22 Aug '11

22 Aug '11

We get called once per DAPM context but only need to run once. When DAPM was serialized this was a series of noops but now it can run in parallel we need to take proper care. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/speyside_wm8962.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c index 753e1c2..3820a6b 100644 --- a/sound/soc/samsung/speyside_wm8962.c +++ b/sound/soc/samsung/speyside_wm8962.c @@ -23,6 +23,9 @@ static int speyside_wm8962_set_bias_level(struct snd_soc_card *card, struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; int ret; + if (dapm->dev != codec_dai->dev) + return 0; + switch (level) { case SND_SOC_BIAS_PREPARE: if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { @@ -57,6 +60,9 @@ static int speyside_wm8962_set_bias_level_post(struct snd_soc_card *card, struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; int ret; + if (dapm->dev != codec_dai->dev) + return 0; + switch (level) { case SND_SOC_BIAS_STANDBY: ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: Add basic WM8962 capture low/high pass filter control
by Mark Brown 22 Aug '11

22 Aug '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 10 ++++++++++ 1 files changed, 10 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 32315ec..eaab0d3 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2173,6 +2173,14 @@ static const char *cap_hpf_mode_text[] = { static const struct soc_enum cap_hpf_mode = SOC_ENUM_SINGLE(WM8962_ADC_DAC_CONTROL_2, 10, 2, cap_hpf_mode_text); + +static const char *cap_lhpf_mode_text[] = { + "LPF", "HPF" +}; + +static const struct soc_enum cap_lhpf_mode = + SOC_ENUM_SINGLE(WM8962_LHPF1, 1, 2, cap_lhpf_mode_text); + static const struct snd_kcontrol_new wm8962_snd_controls[] = { SOC_DOUBLE("Input Mixer Switch", WM8962_INPUT_MIXER_CONTROL_1, 3, 2, 1, 1), @@ -2201,6 +2209,8 @@ SOC_DOUBLE_R("Capture ZC Switch", WM8962_LEFT_INPUT_VOLUME, SOC_SINGLE("Capture HPF Switch", WM8962_ADC_DAC_CONTROL_1, 0, 1, 1), SOC_ENUM("Capture HPF Mode", cap_hpf_mode), SOC_SINGLE("Capture HPF Cutoff", WM8962_ADC_DAC_CONTROL_2, 7, 7, 0), +SOC_SINGLE("Capture LHPF Switch", WM8962_LHPF1, 0, 1, 0), +SOC_ENUM("Capture LHPF Mode", cap_lhpf_mode), SOC_DOUBLE_R_TLV("Sidetone Volume", WM8962_DAC_DSP_MIXING_1, WM8962_DAC_DSP_MIXING_2, 4, 12, 0, st_tlv), -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: Optimise WM8996 no interrupt path
by Mark Brown 22 Aug '11

22 Aug '11

This occurs frequently if we are in edge triggered mode as we must poll the interrupt status register until we get no more interrupts so it's worth the effort - it means we skip writing null acknowledgements to the chip. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index d3277d1..73a1588 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2416,6 +2416,9 @@ static irqreturn_t wm8996_irq(int irq, void *data) } irq_val &= ~snd_soc_read(codec, WM8996_INTERRUPT_STATUS_2_MASK); + if (!irq_val) + return IRQ_NONE; + snd_soc_write(codec, WM8996_INTERRUPT_STATUS_2, irq_val); if (irq_val & (WM8996_DCS_DONE_01_EINT | WM8996_DCS_DONE_23_EINT)) { @@ -2434,10 +2437,7 @@ static irqreturn_t wm8996_irq(int irq, void *data) if (irq_val & WM8996_MICD_EINT) wm8996_micd(codec); - if (irq_val) - return IRQ_HANDLED; - else - return IRQ_NONE; + return IRQ_HANDLED; } static irqreturn_t wm8996_edge_irq(int irq, void *data) -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: Automatically manage WM8996 MICBIAS regulating mode
by Mark Brown 22 Aug '11

22 Aug '11

For non-audio uses like accessory detection we can use a lower quality, unregulated microphone bias, saving a little power. As the hardware can manually enable and disable the biases we can select regulating mode automatically with supply widgets connected to the biases. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 4 ++++ 1 files changed, 4 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 0936ae5..d3277d1 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -982,6 +982,8 @@ SND_SOC_DAPM_SUPPLY_S("Charge Pump", 2, WM8996_CHARGE_PUMP_1, 15, 0, cp_event, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("LDO2", WM8996_POWER_MANAGEMENT_2, 1, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICB1 Audio", WM8996_MICBIAS_1, 4, 1, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICB2 Audio", WM8996_MICBIAS_2, 4, 1, NULL, 0), SND_SOC_DAPM_MICBIAS("MICB2", WM8996_POWER_MANAGEMENT_1, 9, 0), SND_SOC_DAPM_MICBIAS("MICB1", WM8996_POWER_MANAGEMENT_1, 8, 0), @@ -1137,7 +1139,9 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = { { "Charge Pump", NULL, "SYSCLK" }, { "MICB1", NULL, "LDO2" }, + { "MICB1", NULL, "MICB1 Audio" }, { "MICB2", NULL, "LDO2" }, + { "MICB2", NULL, "MICB2 Audio" }, { "IN1L PGA", NULL, "IN2LN" }, { "IN1L PGA", NULL, "IN2LP" }, -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: Fix configuration of WM8996 input enables
by Mark Brown 22 Aug '11

22 Aug '11

There's no need for separate widgets for the enables (as the map already shows). Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 13 ++++--------- 1 files changed, 4 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 0371736..0936ae5 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -988,15 +988,10 @@ SND_SOC_DAPM_MICBIAS("MICB1", WM8996_POWER_MANAGEMENT_1, 8, 0), SND_SOC_DAPM_PGA("IN1L PGA", WM8996_POWER_MANAGEMENT_2, 5, 0, NULL, 0), SND_SOC_DAPM_PGA("IN1R PGA", WM8996_POWER_MANAGEMENT_2, 4, 0, NULL, 0), -SND_SOC_DAPM_MUX("IN1L Mux", SND_SOC_NOPM, 0, 0, &in1_mux), -SND_SOC_DAPM_MUX("IN1R Mux", SND_SOC_NOPM, 0, 0, &in1_mux), -SND_SOC_DAPM_MUX("IN2L Mux", SND_SOC_NOPM, 0, 0, &in2_mux), -SND_SOC_DAPM_MUX("IN2R Mux", SND_SOC_NOPM, 0, 0, &in2_mux), - -SND_SOC_DAPM_PGA("IN1L", WM8996_POWER_MANAGEMENT_7, 2, 0, NULL, 0), -SND_SOC_DAPM_PGA("IN1R", WM8996_POWER_MANAGEMENT_7, 3, 0, NULL, 0), -SND_SOC_DAPM_PGA("IN2L", WM8996_POWER_MANAGEMENT_7, 6, 0, NULL, 0), -SND_SOC_DAPM_PGA("IN2R", WM8996_POWER_MANAGEMENT_7, 7, 0, NULL, 0), +SND_SOC_DAPM_MUX("IN1L Mux", WM8996_POWER_MANAGEMENT_7, 2, 0, &in1_mux), +SND_SOC_DAPM_MUX("IN1R Mux", WM8996_POWER_MANAGEMENT_7, 3, 0, &in1_mux), +SND_SOC_DAPM_MUX("IN2L Mux", WM8996_POWER_MANAGEMENT_7, 6, 0, &in2_mux), +SND_SOC_DAPM_MUX("IN2R Mux", WM8996_POWER_MANAGEMENT_7, 7, 0, &in2_mux), SND_SOC_DAPM_SUPPLY("DMIC2", WM8996_POWER_MANAGEMENT_7, 9, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("DMIC1", WM8996_POWER_MANAGEMENT_7, 8, 0, NULL, 0), -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: WM8996 record paths need AIFCLK
by Mark Brown 22 Aug '11

22 Aug '11

Make AIFCLK supply the record paths otherwise record will not work unless there is a simultaneous playback. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 10 ++++++++++ 1 files changed, 10 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index a44a0e5..0371736 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -1213,6 +1213,16 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = { { "AIF2RX0", NULL, "AIFCLK" }, { "AIF2RX1", NULL, "AIFCLK" }, + { "AIF1TX0", NULL, "AIFCLK" }, + { "AIF1TX1", NULL, "AIFCLK" }, + { "AIF1TX2", NULL, "AIFCLK" }, + { "AIF1TX3", NULL, "AIFCLK" }, + { "AIF1TX4", NULL, "AIFCLK" }, + { "AIF1TX5", NULL, "AIFCLK" }, + + { "AIF2TX0", NULL, "AIFCLK" }, + { "AIF2TX1", NULL, "AIFCLK" }, + { "DSP1RXL", NULL, "SYSDSPCLK" }, { "DSP1RXR", NULL, "SYSDSPCLK" }, { "DSP2RXL", NULL, "SYSDSPCLK" }, -- 1.7.5.4

2 1

[alsa-devel] [PATCH 0/4] ASoC: OMAP4: McPDM: Fix legacy support
by Peter Ujfalusi 22 Aug '11

22 Aug '11

Hello, The OMAP4 McPDM driver in upstream has been broken for some time... This series fixes that, and enables basic audio playback/capture over the mcpdm interface. Two patch in this series has been taken from the "OMAP4/ASoC: New McPDM driver" series (patch 1, and 2) [1]. Since I was on holiday, I did not had time to rewrite the driver rewrite patch, but I feel that fixing the basic audio support for OMAP4 is needed. The aim here is to have working audio in upstream kernel for 3.1, and have the new McPDM driver stack up for the 3.2 release. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2011-July/041687.html Br, Peter --- Peter Ujfalusi (4): OMAP: McPDM: Convert McPDM device to omap_device OMAP4: hwmod: enable mcpdm hwmod device. ASoC: omap-mcpdm: Fix threshold and dma configuration ASoC: OMAP4: McPDM: Convert to hwmod/omap_device arch/arm/mach-omap2/devices.c | 33 ++++++++++++++++++++++++ arch/arm/mach-omap2/omap_hwmod_44xx_data.c | 2 +- arch/arm/plat-omap/devices.c | 36 -------------------------- sound/soc/omap/mcpdm.c | 38 +++++++++++++++------------ sound/soc/omap/mcpdm.h | 1 - sound/soc/omap/omap-mcpdm.c | 19 +++++++++---- sound/soc/omap/sdp4430.c | 2 +- 7 files changed, 69 insertions(+), 62 deletions(-) -- 1.7.6

4 14

[alsa-devel] ALSA underruns for S32_LE format
by kuldeep ghan 22 Aug '11

22 Aug '11

Hi, I am trying to play raw audio as S32_LE format and I am getting underruns. *# aplay /userdata/raw -fS32_LE -r48000 -c2* *Playing raw data '/userdata/raw' : Signed 32 bit Little Endian, Rate 48000 Hz, Stereo* *underrun!!! (at least 0.055 ms long)* *underrun!!! (at least 0.043 ms long)* *underrun!!! (at least 0.040 ms long)* *underrun!!! (at least 0.043 ms long)* *underrun!!! (at least 0.043 ms long)* *underrun!!! (at least 0.043 ms long)* *underrun!!! (at least 0.040 ms long)* I am able play it fine as S16_LE format. (aplay /userdata/raw -fS16_LE -r48000 -c2) I would like to get it working fine S32_LE format. Any help, suggestions to get away with underruns is welcome. Regards, Kuldeep

3 6

[alsa-devel] [PATCH 2/3] ARM: mx28evk: set a initial clock rate for saif
by Dong Aisheng 21 Aug '11

21 Aug '11

Signed-off-by: Dong Aisheng <b29396(a)freescale.com> Cc: Sascha Hauer <s.hauer(a)pengutronix.de> Cc: Wolfram Sang <w.sang(a)pengutronix.de> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg(a)ti.com> --- Note this patch still needs another patch from Wolfram to work. I send out it in this series as [PATCH 3/3] arm: mxs: disable clock-gates when setting saif-clocks As discussed with Wolfram, that patch may still have some later work to do. However, we just send out it for testers to work first. --- arch/arm/mach-mxs/clock-mx28.c | 9 +++++++++ 1 files changed, 9 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-mxs/clock-mx28.c b/arch/arm/mach-mxs/clock-mx28.c index f589c6c..2a2db65 100644 --- a/arch/arm/mach-mxs/clock-mx28.c +++ b/arch/arm/mach-mxs/clock-mx28.c @@ -788,6 +788,15 @@ int __init mx28_clocks_init(void) clk_set_parent(&saif0_clk, &pll0_clk); clk_set_parent(&saif1_clk, &pll0_clk); + /* + * Set an initial clk rate for saif's internal logic to work properly, + * this is especially for the saif working on EXTMASTER mode that who + * uses other saif's BITCLK&LRCLK but it still needs a basic clk which + * should be bigger enough for its internal logic. + */ + clk_set_rate(&saif0_clk, 24000000); + clk_set_rate(&saif1_clk, 24000000); + clkdev_add_table(lookups, ARRAY_SIZE(lookups)); mxs_timer_init(&clk32k_clk, MX28_INT_TIMER0); -- 1.7.0.4

1 0

[alsa-devel] [PATCH 1/3] ARM: mx28evk: add platform data for saif
by Dong Aisheng 21 Aug '11

21 Aug '11

This is for supporting saif record function. Signed-off-by: Dong Aisheng <b29396(a)freescale.com> Cc: Sascha Hauer <s.hauer(a)pengutronix.de> Cc: Wolfram Sang <w.sang(a)pengutronix.de> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg(a)ti.com> --- arch/arm/mach-mxs/clock-mx28.c | 9 +++++++++ arch/arm/mach-mxs/devices-mx28.h | 4 +++- arch/arm/mach-mxs/devices/platform-mxs-saif.c | 5 +++-- arch/arm/mach-mxs/include/mach/devices-common.h | 4 +++- arch/arm/mach-mxs/include/mach/digctl.h | 18 ++++++++++++++++++ arch/arm/mach-mxs/mach-mx28evk.c | 16 ++++++++++++++-- 6 files changed, 50 insertions(+), 6 deletions(-) diff --git a/arch/arm/mach-mxs/clock-mx28.c b/arch/arm/mach-mxs/clock-mx28.c index 7954013..f589c6c 100644 --- a/arch/arm/mach-mxs/clock-mx28.c +++ b/arch/arm/mach-mxs/clock-mx28.c @@ -44,6 +44,15 @@ static struct clk saif0_clk; static struct clk saif1_clk; static struct clk clk32k_clk; +int mxs_saif_clkmux_select(unsigned int clkmux) +{ + if (clkmux > 0x3) + return -EINVAL; + + __raw_writel(clkmux << 10, DIGCTRL_BASE_ADDR + MXS_SET_ADDR); + return 0; +} + static int _raw_clk_enable(struct clk *clk) { u32 reg; diff --git a/arch/arm/mach-mxs/devices-mx28.h b/arch/arm/mach-mxs/devices-mx28.h index c888710..40be649 100644 --- a/arch/arm/mach-mxs/devices-mx28.h +++ b/arch/arm/mach-mxs/devices-mx28.h @@ -47,6 +47,8 @@ struct platform_device *__init mx28_add_mxsfb( const struct mxsfb_platform_data *pdata); extern const struct mxs_saif_data mx28_saif_data[] __initconst; -#define mx28_add_saif(id) mxs_add_saif(&mx28_saif_data[id]) +#define mx28_add_saif(id, pdata) \ + mxs_add_saif(&mx28_saif_data[id], pdata) + struct platform_device *__init mx28_add_rtc_stmp3xxx(void); diff --git a/arch/arm/mach-mxs/devices/platform-mxs-saif.c b/arch/arm/mach-mxs/devices/platform-mxs-saif.c index 1ec965e..f6e3a60 100644 --- a/arch/arm/mach-mxs/devices/platform-mxs-saif.c +++ b/arch/arm/mach-mxs/devices/platform-mxs-saif.c @@ -32,7 +32,8 @@ const struct mxs_saif_data mx28_saif_data[] __initconst = { }; #endif -struct platform_device *__init mxs_add_saif(const struct mxs_saif_data *data) +struct platform_device *__init mxs_add_saif(const struct mxs_saif_data *data, + const struct mxs_saif_platform_data *pdata) { struct resource res[] = { { @@ -56,5 +57,5 @@ struct platform_device *__init mxs_add_saif(const struct mxs_saif_data *data) }; return mxs_add_platform_device("mxs-saif", data->id, res, - ARRAY_SIZE(res), NULL, 0); + ARRAY_SIZE(res), pdata, sizeof(*pdata)); } diff --git a/arch/arm/mach-mxs/include/mach/devices-common.h b/arch/arm/mach-mxs/include/mach/devices-common.h index a8080f4..dc369c1 100644 --- a/arch/arm/mach-mxs/include/mach/devices-common.h +++ b/arch/arm/mach-mxs/include/mach/devices-common.h @@ -94,6 +94,7 @@ struct platform_device *__init mxs_add_mxs_pwm( resource_size_t iobase, int id); /* saif */ +#include <sound/saif.h> struct mxs_saif_data { int id; resource_size_t iobase; @@ -103,4 +104,5 @@ struct mxs_saif_data { }; struct platform_device *__init mxs_add_saif( - const struct mxs_saif_data *data); + const struct mxs_saif_data *data, + const struct mxs_saif_platform_data *pdata); diff --git a/arch/arm/mach-mxs/include/mach/digctl.h b/arch/arm/mach-mxs/include/mach/digctl.h new file mode 100644 index 0000000..6231e10 --- /dev/null +++ b/arch/arm/mach-mxs/include/mach/digctl.h @@ -0,0 +1,18 @@ +/* + * 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 __MACH_DIGCTL_H__ +#define __MACH_DIGCTL_H__ + +/* MXS DIGCTL SAIF CLKMUX */ +#define MXS_DIGCTL_SAIF_CLKMUX_DIRECT 0x0 +#define MXS_DIGCTL_SAIF_CLKMUX_CROSSINPUT 0x1 +#define MXS_DIGCTL_SAIF_CLKMUX_EXTMSTR0 0x2 +#define MXS_DIGCTL_SAIF_CLKMUX_EXTMSTR1 0x3 + +#endif diff --git a/arch/arm/mach-mxs/mach-mx28evk.c b/arch/arm/mach-mxs/mach-mx28evk.c index 4a3cca3..f3ef9e9 100644 --- a/arch/arm/mach-mxs/mach-mx28evk.c +++ b/arch/arm/mach-mxs/mach-mx28evk.c @@ -28,6 +28,7 @@ #include <mach/common.h> #include <mach/iomux-mx28.h> +#include <mach/digctl.h> #include "devices-mx28.h" @@ -417,6 +418,17 @@ static void __init mx28evk_add_regulators(void) static void __init mx28evk_add_regulators(void) {} #endif +extern int mxs_saif_clkmux_select(unsigned int clkmux); +static int mx28evk_mxs_saif_pinit(void) +{ + return mxs_saif_clkmux_select(MXS_DIGCTL_SAIF_CLKMUX_EXTMSTR0); +} + +struct mxs_saif_platform_data mx28evk_mxs_saif_pdata __initdata = { + .init = mx28evk_mxs_saif_pinit, + .clkmux = MXS_DIGCTL_SAIF_CLKMUX_EXTMSTR0, +}; + static void __init mx28evk_init(void) { int ret; @@ -457,8 +469,8 @@ static void __init mx28evk_init(void) mx28_add_mxsfb(&mx28evk_mxsfb_pdata); - mx28_add_saif(0); - mx28_add_saif(1); + mx28_add_saif(0, &mx28evk_mxs_saif_pdata); + mx28_add_saif(1, &mx28evk_mxs_saif_pdata); mx28_add_mxs_i2c(0); i2c_register_board_info(0, mxs_i2c0_board_info, -- 1.7.0.4

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[alsa-devel] ASoC updates for 3.1
by Mark Brown 21 Aug '11

21 Aug '11

The following changes since commit f6b864a9071e21186476910613ec9913b56067a2: ASoC: Fix compile warning in wm8750.c (2011-08-12 18:22:10 +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 The biggest change here is cleaning up the ad193x mess - there appear to have been quite a few issues on top of the entertainment with the register I/O. Axel Lin (2): ASoC: soc-jack: Fix checking return value of request_any_context_irq ASoC: sta32x: Fix a memory leak if snd_soc_register_codec fails Jarkko Nikula (1): ASoC: omap: Fix build errors in ams-delta Mark Brown (1): ASoC: Move WM8962 CLKREG_OVD earlier Sascha Hauer (1): ASoC: Fix check for symmetric rate enforcement Scott Jiang (5): ASoC: ad193x: fix registers definition ASoC: ad193x: fix dac word len setting ASoC: ad193x: fix system clock ASoC: Add spi hw read function for 16 addr 8 data mode for ad193x fix ASoC: ad193x: remove cache support Vasily Khoruzhick (2): ASoC: h1940: Fix compilation error due to missing header ASoC: rx1950: Fix compilation error due to missing header sound/soc/blackfin/bf5xx-ad193x.c | 2 +- sound/soc/codecs/ad193x.c | 11 ++--------- sound/soc/codecs/ad193x.h | 5 +++-- sound/soc/codecs/sta32x.c | 1 + sound/soc/codecs/wm8962.c | 8 ++++---- sound/soc/omap/ams-delta.c | 6 ++++-- sound/soc/samsung/h1940_uda1380.c | 1 + sound/soc/samsung/rx1950_uda1380.c | 1 + sound/soc/soc-io.c | 23 +++++++++++++++++++++++ sound/soc/soc-jack.c | 2 +- sound/soc/soc-pcm.c | 3 +++ 11 files changed, 44 insertions(+), 19 deletions(-)

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[alsa-devel] ASoC updates for 3.2
by Mark Brown 21 Aug '11

21 Aug '11

The following changes since commit c18eee31812d42ecd0aa5c39d21d41c15b30eaab: ASoC: Add bitfield definitions for WM8958 MICBIAS registers (2011-08-12 14:23:04 +0900) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.2 Axel Lin (10): ASoC: soc-jack: Fix checking return value of request_any_context_irq ASoC: sta32x: Fix a memory leak if snd_soc_register_codec fails ASoC: s6000-pcm: remove unused variable 'dai' ASoC: soc-cache: Remove unneeded codec_drv pointer variable in snd_soc_flat_cache_init ASoC: soc-cache: Remove unneeded codec_drv pointer variable in snd_soc_lzo_get_blksize ASoC: soc-pcm: Remove unused global mutex ASoC: nuc900-pcm: remove unused variable 'dai' ASoC: sta32x: shortcut the for loop to get ir and mcs ASoC: Remove unreachable code in au1xac97c_drvprobe and au1xi2s_drvprobe ASoC: sta32x: Move resource allocation and release to the corresponding callback functions Jarkko Nikula (1): ASoC: omap: Fix build errors in ams-delta Lars-Peter Clausen (3): ASoC: DAPM: Allow multiple mixer sources to be routed via the same switch ASoC: Add ADAU1373 codec support ASoC: Blackfin: ADAU1373 eval board support Mark Brown (25): ASoC: Add device tree binding for WM8510 ASoC: Add device tree binding for WM8523 ASoC: Add device tree support for WM8580 ASoC: Add device tree binding for WM8711 ASoC: Add device tree binding for WM8728 ASoC: Add device tree binding for WM8737 ASoC: Add device tree binding for WM8750 ASoC: Add device tree binding for WM8753 ASoC: Add VMID widget for wm_hubs devices ASoC: Disable WM8994 VMID for digital only paths ASoC: Move WM8962 CLKREG_OVD earlier ASoC: Remove redundant -codec from WM8750 Merge branch 'for-3.1' into for-3.2 ASoC: Support !CONFIG_REGMAP builds Merge branch 'for-3.1' into for-3.2 Merge branch 'for-3.1' into for-3.2 ASoC: Add device tree binding for WM8741 ASoC: Correct revision display for WM1250-EV1 module ASoC: Disable pulls on WM8994 AIF2 when starting it ASoC: Run Speyside WM8962 at 512fs ASoC: Add WM8958 noise gate support ASoC: Implement WM8994 thermal warning and shutdown interrupt support ASoC: Fix backport of WM8994 thermal warning Merge branch 'for-3.1' into for-3.2 Merge branch 'for-3.1' into for-3.2 Sascha Hauer (1): ASoC: Fix check for symmetric rate enforcement Scott Jiang (5): ASoC: ad193x: fix registers definition ASoC: ad193x: fix dac word len setting ASoC: ad193x: fix system clock ASoC: Add spi hw read function for 16 addr 8 data mode for ad193x fix ASoC: ad193x: remove cache support Takashi Iwai (1): ASoC: Fix compile warning in wm8750.c Timur Tabi (2): ASoC: claim the IRQ when the fsl_ssi device is probed, not opened ASoC: fsl: fix build warning in fsl_dma Vasily Khoruzhick (2): ASoC: h1940: Fix compilation error due to missing header ASoC: rx1950: Fix compilation error due to missing header Documentation/devicetree/bindings/sound/wm8510.txt | 18 + Documentation/devicetree/bindings/sound/wm8523.txt | 16 + Documentation/devicetree/bindings/sound/wm8580.txt | 16 + Documentation/devicetree/bindings/sound/wm8711.txt | 18 + Documentation/devicetree/bindings/sound/wm8728.txt | 18 + Documentation/devicetree/bindings/sound/wm8737.txt | 18 + Documentation/devicetree/bindings/sound/wm8741.txt | 18 + Documentation/devicetree/bindings/sound/wm8750.txt | 18 + Documentation/devicetree/bindings/sound/wm8753.txt | 18 + include/linux/mfd/wm8994/registers.h | 45 + include/sound/adau1373.h | 34 + sound/soc/au1x/ac97c.c | 2 - sound/soc/au1x/i2sc.c | 1 - sound/soc/blackfin/Kconfig | 13 + sound/soc/blackfin/Makefile | 2 + sound/soc/blackfin/bf5xx-ad193x.c | 2 +- sound/soc/blackfin/bfin-eval-adau1373.c | 202 +++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ad193x.c | 11 +- sound/soc/codecs/ad193x.h | 5 +- sound/soc/codecs/adau1373.c | 1414 ++++++++++++++++++++ sound/soc/codecs/adau1373.h | 29 + sound/soc/codecs/sta32x.c | 22 +- sound/soc/codecs/wm1250-ev1.c | 2 +- sound/soc/codecs/wm8510.c | 8 + sound/soc/codecs/wm8523.c | 7 + sound/soc/codecs/wm8580.c | 7 + sound/soc/codecs/wm8711.c | 9 + sound/soc/codecs/wm8728.c | 9 + sound/soc/codecs/wm8737.c | 10 + sound/soc/codecs/wm8741.c | 9 + sound/soc/codecs/wm8750.c | 16 +- sound/soc/codecs/wm8753.c | 9 + sound/soc/codecs/wm8962.c | 8 +- sound/soc/codecs/wm8993.c | 4 + sound/soc/codecs/wm8994-tables.c | 12 +- sound/soc/codecs/wm8994.c | 264 +++- sound/soc/codecs/wm8994.h | 2 + sound/soc/codecs/wm_hubs.c | 16 + sound/soc/fsl/fsl_dma.c | 1 - sound/soc/fsl/fsl_ssi.c | 61 +- sound/soc/nuc900/nuc900-pcm.c | 1 - sound/soc/omap/ams-delta.c | 6 +- sound/soc/pxa/spitz.c | 2 +- sound/soc/pxa/z2.c | 2 +- sound/soc/s6000/s6000-pcm.c | 1 - sound/soc/samsung/h1940_uda1380.c | 1 + sound/soc/samsung/jive_wm8750.c | 2 +- sound/soc/samsung/rx1950_uda1380.c | 1 + sound/soc/samsung/smartq_wm8987.c | 2 +- sound/soc/samsung/speyside_wm8962.c | 4 +- sound/soc/soc-cache.c | 7 - sound/soc/soc-dapm.c | 6 +- sound/soc/soc-io.c | 11 +- sound/soc/soc-jack.c | 2 +- sound/soc/soc-pcm.c | 5 +- 57 files changed, 2309 insertions(+), 144 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/wm8510.txt create mode 100644 Documentation/devicetree/bindings/sound/wm8523.txt create mode 100644 Documentation/devicetree/bindings/sound/wm8580.txt create mode 100644 Documentation/devicetree/bindings/sound/wm8711.txt create mode 100644 Documentation/devicetree/bindings/sound/wm8728.txt create mode 100644 Documentation/devicetree/bindings/sound/wm8737.txt create mode 100644 Documentation/devicetree/bindings/sound/wm8741.txt create mode 100644 Documentation/devicetree/bindings/sound/wm8750.txt create mode 100644 Documentation/devicetree/bindings/sound/wm8753.txt create mode 100644 include/sound/adau1373.h create mode 100644 sound/soc/blackfin/bfin-eval-adau1373.c create mode 100644 sound/soc/codecs/adau1373.c create mode 100644 sound/soc/codecs/adau1373.h

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Re: [alsa-devel] snd-ml403-ac97cr ALSA DRIVER
by Paul Menzel 20 Aug '11

20 Aug '11

Dear Ari, Am Samstag, den 20.08.2011, 20:04 +0200 schrieb Ari Kaspari: > Ok thank you, > i did not know. > > I loaded my ml403-ac97cr modul with modprobe. > but in my alsa files are no sound cards detected. > > i tried to use mpg123 do "debug" and it said no sound card '0' available > > Any idea? I have no idea and I have no crystal ball. Since you top posted and only replied to me personally that indicates to me that you in fact did not read my reply and the links I included. I am sorry most of the people here try to help others in their free time and I am not going to do the simple work for you to provide the information *you* need to provide and I send you the links for. Do your homework first and I am sorry at least I will not and cannot help you further. If you want others to do the work for you, please get a support contract you have to pay for and the problem will be solved for you. Thanks, Paul

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[alsa-devel] snd-ml403-ac97cr ALSA DRIVER
by Ari Kaspari 20 Aug '11

20 Aug '11

Hello, has anyone used the snd-ml403-ac97cr ALSA DRIVER with success? Could anyone play Audio samples with this driver? In my project the driver don't work. I hope i hear from you Ari

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[alsa-devel] [PATCH 4/4] sound/aoa/fabrics/layout.c: remove unneeded kfree
by Julia Lawall 20 Aug '11

20 Aug '11

From: Julia Lawall <julia(a)diku.dk> The label outnodev is only used when kzalloc has not yet taken place or has failed, so there is no need for the call for kfree under this label. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ identifier x; expression E1!=0,E2,E3,E4; statement S; iterator I; @@ ( if (...) { ... when != kfree(x) when != x = E3 when != E3 = x * return ...; } ... when != x = E2 when != I(...,x,...) S if (...) { ... when != x = E4 kfree(x); ... return ...; } ) // </smpl> Signed-off-by: Julia Lawall <julia(a)diku.dk> --- sound/aoa/fabrics/layout.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c index 3fd1a7e..552b97a 100644 --- a/sound/aoa/fabrics/layout.c +++ b/sound/aoa/fabrics/layout.c @@ -1073,10 +1073,10 @@ static int aoa_fabric_layout_probe(struct soundbus_dev *sdev) sdev->pcmid = -1; list_del(&ldev->list); layouts_list_items--; + kfree(ldev); outnodev: of_node_put(sound); layout_device = NULL; - kfree(ldev); return -ENODEV; }

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Re: [alsa-devel] [PATCH] Sound, Cirrus: Remove redundant check
by Tabi Timur-B04825 20 Aug '11

20 Aug '11

On Fri, Aug 19, 2011 at 6:00 PM, Jesper Juhl <jj(a)chaosbits.net> wrote: > The variable 'idxp' is a pointer to 'unsigned int', so the value > obtained when dereferencing it is always greater than or equal to > 0. Testing this is redundant. snd_hda_get_conn_index() can return a negative number, so maybe *idxp should be a pointer to a signed int? -- Timur Tabi Linux kernel developer at Freescale

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Re: [alsa-devel] USB Audio questions
by Clemens Ladisch 20 Aug '11

20 Aug '11

Pierre-Louis Bossart wrote: > Any idea on how this frame number can be read, With usb_get_current_frame_number(). This is not a technical problem; the problem is that nobody has found the time to implement this in the audio driver. Regards, Clemens

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[alsa-devel] Multiple asoc cards issue?
by Le Gall, Louis 19 Aug '11

19 Aug '11

Hello We are trying to add a new ASOC card in our build that already contains an ASOC card. We facing some issue when the second card is registered (calling snd_soc_register_card () ). We can see in the log when playing a sound on the first card that second card operations are called. In fact, the registration of second card overwrites the operations of the structure soc_pcm_ops of the first card (with the second card operations). Looking at the soc-core.c code, this structure is static so that explains the overwrite. How can we register several ASOC cards on our platform in that case? Is it a bug or a limitation? Thanks, Louis First card = First card registered during boot process Second card = second card that is registred during the boot process. --------------------------------------------------------------------- Intel Corporation SAS (French simplified joint stock company) Registered headquarters: "Les Montalets"- 2, rue de Paris, 92196 Meudon Cedex, France Registration Number: 302 456 199 R.C.S. NANTERRE Capital: 4,572,000 Euros This e-mail and any attachments may contain confidential material for the sole use of the intended recipient(s). Any review or distribution by others is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies.

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[alsa-devel] asoc: no valid backend routes for PCM: SDP4430 Media
by hong zhang 19 Aug '11

19 Aug '11

List, After capturing data from DMIC on OMAP4 blaze board with kernel 2.6.38 and save wav data to 38.wav. But aplay complains "no valid backend routes". See following, $aplay -f dat 38.wav [ 157.167724] asoc: no valid backend routes for PCM: SDP4430 Media aplay: main:654: audio open error: Invalid argument $ uname -a Linux ubuntu 2.6.38.4+ #58 SMP PREEMPT Fri Jun 3 09:44:19 PDT 2011 armv7l GNU/Linux Does ALSA add new functionality of backend? Thanks. ---henry

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[alsa-devel] [PATCH] alsaucm: exit when user wants 'help' and 'version'
by Lu Guanqun 19 Aug '11

19 Aug '11

When user specifies option 'help' or 'version', the program exists immediately. This is more expected way than the current one. Signed-off-by: Lu Guanqun <guanqun.lu(a)intel.com> --- alsaucm/usecase.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/alsaucm/usecase.c b/alsaucm/usecase.c index 1c94680..14a4be0 100644 --- a/alsaucm/usecase.c +++ b/alsaucm/usecase.c @@ -395,10 +395,10 @@ int main(int argc, char *argv[]) switch (c) { case 'h': dump_help(context); - break; + my_exit(context, EXIT_SUCCESS); case OPT_VERSION: printf("%s: version " SND_UTIL_VERSION_STR "\n", command); - break; + my_exit(context, EXIT_SUCCESS); case 'c': if (context->card) free(context->card);

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[PATCH] alsa-plugins: Pulse: only underrun if no more data has been written
by David Henningsson 19 Aug '11

19 Aug '11

If more data has already been written after the underrun, the underrun will automatically end and therefore we should not report it or restart the stream. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- configure.in | 7 +++++++ pulse/pcm_pulse.c | 28 ++++++++++++++++++++++++++-- 2 files changed, 33 insertions(+), 2 deletions(-) diff --git a/configure.in b/configure.in index ccf59ba..f6886b1 100644 --- a/configure.in +++ b/configure.in @@ -31,8 +31,14 @@ AC_ARG_ENABLE([pulseaudio], if test "x$enable_pulseaudio" != "xno"; then PKG_CHECK_MODULES(pulseaudio, [libpulse >= 0.9.11], [HAVE_PULSE=yes], [HAVE_PULSE=no]) + PKG_CHECK_MODULES(pulseaudio_099, [libpulse >= 0.99.1], + [HAVE_PULSE_UNDERRUN_INDEX=yes], [HAVE_PULSE_UNDERRUN_INDEX=no]) + fi AM_CONDITIONAL(HAVE_PULSE, test x$HAVE_PULSE = xyes) +if test "$HAVE_PULSE_UNDERRUN_INDEX" = "yes"; then + AC_DEFINE([HAVE_PULSE_UNDERRUN_INDEX], 1, Pulseaudio underrun index available) +fi AC_ARG_ENABLE([samplerate], AS_HELP_STRING([--disable-samplerate], [Disable building of samplerate plugin])) @@ -185,6 +191,7 @@ echo "Pulseaudio plugin: $HAVE_PULSE" if test "$HAVE_PULSE" = "yes"; then echo " pulseaudio_CFLAGS: $pulseaudio_CFLAGS" echo " pulseaudio_LIBS: $pulseaudio_LIBS" + echo " pulseaudio underrun index: $HAVE_PULSE_UNDERRUN_INDEX" fi echo "Samplerate plugin: $HAVE_SAMPLERATE" if test "$HAVE_SAMPLERATE" = "yes"; then diff --git a/pulse/pcm_pulse.c b/pulse/pcm_pulse.c index d6c6792..61f1c0c 100644 --- a/pulse/pcm_pulse.c +++ b/pulse/pcm_pulse.c @@ -26,6 +26,10 @@ #include <alsa/asoundlib.h> #include <alsa/pcm_external.h> +#ifdef HAVE_CONFIG_H +#include <config.h> +#endif + #include "pulse.h" typedef struct snd_pcm_pulse { @@ -39,9 +43,10 @@ typedef struct snd_pcm_pulse { size_t last_size; size_t ptr; int underrun; - int handle_underrun; + int handle_underrun; /* can be 0=never, 1=always or 2,3=only if more data has not been written */ size_t offset; + int64_t written; pa_stream *stream; @@ -459,6 +464,7 @@ static snd_pcm_sframes_t pulse_write(snd_pcm_ioplug_t * io, } /* Make sure the buffer pointer is in sync */ + pcm->written += writebytes; pcm->last_size -= writebytes; ret = update_ptr(pcm); if (ret < 0) @@ -594,7 +600,12 @@ static void stream_underrun_cb(pa_stream * p, void *userdata) if (!pcm->p) return; - pcm->underrun = 1; + if (pcm->handle_underrun == 1 +#ifdef HAVE_PULSE_UNDERRUN_INDEX + || pcm->written <= pa_stream_get_underflow_index(p) +#endif + ) + pcm->underrun = 1; } static void stream_latency_cb(pa_stream *p, void *userdata) { @@ -691,6 +702,7 @@ static int pulse_prepare(snd_pcm_ioplug_t * io) goto finish; } + pcm->written = 0; pa_stream_set_state_callback(pcm->stream, stream_state_cb, pcm); pa_stream_set_latency_update_callback(pcm->stream, stream_latency_cb, pcm); @@ -983,7 +995,11 @@ SND_PCM_PLUGIN_DEFINE_FUNC(pulse) const char *server = NULL; const char *device = NULL; const char *fallback_name = NULL; +#ifdef HAVE_PULSE_UNDERRUN_INDEX + int handle_underrun = 2; +#else int handle_underrun = 0; +#endif int err; snd_pcm_pulse_t *pcm; @@ -1017,6 +1033,14 @@ SND_PCM_PLUGIN_DEFINE_FUNC(pulse) handle_underrun = err; continue; } + if (strcmp(id, "handle_underrun_detect") == 0) { + if ((err = snd_config_get_bool(n)) < 0) { + SNDERR("Invalid value for %s", id); + return -EINVAL; + } + handle_underrun = err ? handle_underrun | 2 : handle_underrun & ~2; + continue; + } if (strcmp(id, "fallback") == 0) { if (snd_config_get_string(n, &fallback_name) < 0) { SNDERR("Invalid value for %s", id); -- 1.7.5.4 --------------040503050706040000000002--

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[alsa-devel] Getting played tsms in ALSA
by Irfan Shaikh 19 Aug '11

19 Aug '11

Hello all At setup time, when we create alsa handle and get the pcm_status object from it. /******************************/ snd_pcm_status *status; snd_pcm_status_alloca(&status); At run time: snd_timestamp_t timestamp = {0}; snd_pcm_status(pcm_handle, status); snd_pcm_status_get_tstamp(status, &timestamp); print timestamp.tv_sec and timestamp.tv_usec /******************************/ Here when we start first instance, timestamp.tv_sec goes incrementing as we play as expected. But we when shutdown (destroy alsa handle, re-init) , the timestamps increase from the same point. i.e if we stopped playback at time x secs, the next instance starts showing the time from x secs. The time reported by snd_pcm_status_get_tstamp() resets after we reboot the board. Another observation is that when we have two instances. We start the first instance , pause it after x secs. And then start the next instance, the time given by ALSA again starts at x secs. I think the snd_pcm_get_tstamp should be handle based but it seems it's not the case. How can we reset the timestamp values ? Regards SASKEN BUSINESS DISCLAIMER: This message may contain confidential, proprietary or legally privileged information. In case you are not the original intended Recipient of the message, you must not, directly or indirectly, use, disclose, distribute, print, or copy any part of this message and you are requested to delete it and inform the sender. Any views expressed in this message are those of the individual sender unless otherwise stated. Nothing contained in this message shall be construed as an offer or acceptance of any offer by Sasken Communication Technologies Limited ("Sasken") unless sent with that express intent and with due authority of Sasken. Sasken has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email. Read Disclaimer at http://www.sasken.com/extras/mail_disclaimer.html

1 0

[alsa-devel] [PATCH 1/2] ucm: fix seg fault in execute_cset()
by Lu Guanqun 19 Aug '11

19 Aug '11

When there's no space in 'cset', 'pos' gets NULL, dereferencing 'pos' gets a seg fault. We need to be more robust. Signed-off-by: Lu Guanqun <guanqun.lu(a)intel.com> --- src/ucm/main.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/src/ucm/main.c b/src/ucm/main.c index 8e9a85d..9ecbe43 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -196,7 +196,8 @@ static int execute_cset(snd_ctl_t *ctl, char *cset) goto __fail; err = 0; __fail: - *pos = ' '; + if (pos != NULL) + *pos = ' '; if (id != NULL) free(id);

2 2

[alsa-devel] M-Audio FTU remaining mixer controls and external clocking support.
by Grant Diffey 19 Aug '11

19 Aug '11

Daniel, I remember you saying you'd added the remaining controls for the ftu and had worked out how to enable external clock support? Was that right or am I just remembering optimistically, Additionally have you had any thoughts on clemens outline of how the api between the pcm and usb stuff should work? Grant.

2 1

[alsa-devel] Audio:suspend to ram issue
by Rajeev kumar 19 Aug '11

19 Aug '11

Hi Mark I am not sure about the behavior in case of audio suspend/resume. I tested the audio driver with aplay and then suspended to ram (using echo mem > /sys/power/state). After pressing wake to resume, the audio does not play unless aplay is closed and re-run. Do we expect the audio suspend/resume to work? In the platform trigger part I am doing nothing and in case of cpu(i2s) , just disabling the i2s. Best Regards Rajeev

2 5

Re: [alsa-devel] hda-intel and runtime power-management
by Takashi Iwai 19 Aug '11

19 Aug '11

At Mon, 15 Aug 2011 16:04:46 -0500, Pierre-Louis Bossart wrote: > > > Another approach would be to set the stream to mute when paused and let > > the codec do the amp power check which will power it down if needed. > > What do you think? > > PulseAudio and other equivalent layers don't pause the output. They close > the ALSA sink after a timeout and reopen it when the sink/Hal state change. > You may want to avoid working your tail off with a seemingly smart way of > using the paused state, if in the end it's not that useful... Indeed, there are only few apps that use the pause operation, so it won't a big win in comparison to the added code complexity. thanks, Takashi

1 0

Re: [alsa-devel] [PATCH 1/1] Sound, echoaudio: Place echoaudio drivers in submenu
by Takashi Iwai 19 Aug '11

19 Aug '11

At Thu, 18 Aug 2011 10:11:12 -0700, John Daiker wrote: > > Put all the echoaudio boards into their own Kconfig submneu. They used to be > listed individually. Because they are indeed individual drivers :) I don't mind much to include such a patch, but OTOH, people also don't like to add yet-another-config item. Also default=y isn't always preferred... thanks, Takashi > > Signed-off-by: John Daiker <daikerjohn(a)gmail.com> > --- > sound/pci/Kconfig | 25 +++++++++++++++++++++++++ > 1 files changed, 25 insertions(+), 0 deletions(-) > > diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig > index 8816804..9344274 100644 > --- a/sound/pci/Kconfig > +++ b/sound/pci/Kconfig > @@ -312,8 +312,18 @@ config SND_CTXFI > To compile this driver as a module, choose M here: the module > will be called snd-ctxfi. > > +config SND_ECHOAUDIO > + tristate "Echoaudio Support" > + default y > + select SND_PCM > + help > + Support for EchoAudio based devices. > + > +if SND_ECHOAUDIO > + > config SND_DARLA20 > tristate "(Echoaudio) Darla20" > + depends on SND_ECHOAUDIO > select FW_LOADER > select SND_PCM > help > @@ -324,6 +334,7 @@ config SND_DARLA20 > > config SND_GINA20 > tristate "(Echoaudio) Gina20" > + depends on SND_ECHOAUDIO > select FW_LOADER > select SND_PCM > help > @@ -334,6 +345,7 @@ config SND_GINA20 > > config SND_LAYLA20 > tristate "(Echoaudio) Layla20" > + depends on SND_ECHOAUDIO > select FW_LOADER > select SND_RAWMIDI > select SND_PCM > @@ -345,6 +357,7 @@ config SND_LAYLA20 > > config SND_DARLA24 > tristate "(Echoaudio) Darla24" > + depends on SND_ECHOAUDIO > select FW_LOADER > select SND_PCM > help > @@ -355,6 +368,7 @@ config SND_DARLA24 > > config SND_GINA24 > tristate "(Echoaudio) Gina24" > + depends on SND_ECHOAUDIO > select FW_LOADER > select SND_PCM > help > @@ -365,6 +379,7 @@ config SND_GINA24 > > config SND_LAYLA24 > tristate "(Echoaudio) Layla24" > + depends on SND_ECHOAUDIO > select FW_LOADER > select SND_RAWMIDI > select SND_PCM > @@ -376,6 +391,7 @@ config SND_LAYLA24 > > config SND_MONA > tristate "(Echoaudio) Mona" > + depends on SND_ECHOAUDIO > select FW_LOADER > select SND_RAWMIDI > select SND_PCM > @@ -387,6 +403,7 @@ config SND_MONA > > config SND_MIA > tristate "(Echoaudio) Mia" > + depends on SND_ECHOAUDIO > select FW_LOADER > select SND_RAWMIDI > select SND_PCM > @@ -398,6 +415,7 @@ config SND_MIA > > config SND_ECHO3G > tristate "(Echoaudio) 3G cards" > + depends on SND_ECHOAUDIO > select FW_LOADER > select SND_RAWMIDI > select SND_PCM > @@ -409,6 +427,7 @@ config SND_ECHO3G > > config SND_INDIGO > tristate "(Echoaudio) Indigo" > + depends on SND_ECHOAUDIO > select FW_LOADER > select SND_PCM > help > @@ -419,6 +438,7 @@ config SND_INDIGO > > config SND_INDIGOIO > tristate "(Echoaudio) Indigo IO" > + depends on SND_ECHOAUDIO > select FW_LOADER > select SND_PCM > help > @@ -429,6 +449,7 @@ config SND_INDIGOIO > > config SND_INDIGODJ > tristate "(Echoaudio) Indigo DJ" > + depends on SND_ECHOAUDIO > select FW_LOADER > select SND_PCM > help > @@ -439,6 +460,7 @@ config SND_INDIGODJ > > config SND_INDIGOIOX > tristate "(Echoaudio) Indigo IOx" > + depends on SND_ECHOAUDIO > select FW_LOADER > select SND_PCM > help > @@ -449,6 +471,7 @@ config SND_INDIGOIOX > > config SND_INDIGODJX > tristate "(Echoaudio) Indigo DJx" > + depends on SND_ECHOAUDIO > select FW_LOADER > select SND_PCM > help > @@ -457,6 +480,8 @@ config SND_INDIGODJX > To compile this driver as a module, choose M here: the module > will be called snd-indigodjx > > +endif > + > config SND_EMU10K1 > tristate "Emu10k1 (SB Live!, Audigy, E-mu APS)" > select FW_LOADER > -- > 1.7.4.1 >

1 0

[alsa-devel] [PATCH] Prefer kernel-versioned firmware directory.
by linux＠audioscience.com 19 Aug '11

19 Aug '11

From: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> Different kernel versions may have different driver versions installed, which in turn require different firmware versions. If /lib/firmware/$(uname -r) exists, use it in preference to the generic /lib/firmware. Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> --- configure.in | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/configure.in b/configure.in index c278987..ac9ee52 100644 --- a/configure.in +++ b/configure.in @@ -33,7 +33,9 @@ AM_CONDITIONAL(BUILD_FW, test "$buildfw" = "yes") AC_MSG_CHECKING(firmware installation directory) # where to put the firmware. If none of these is right, can specify where # --with-hotplug-dir -if test -d /lib/firmware ; then +if test -d /lib/firmware/$(uname -r) ; then + detected_fwdir="/lib/firmware/$(uname -r)" +elif test -d /lib/firmware ; then detected_fwdir="/lib/firmware" elif test -d /lib/hotplug/firmware ; then detected_fwdir="/lib/hotplug/firmware" -- 1.7.0.4

3 3

Re: [alsa-devel] Lost dB volumes on USB headset between 2.6.38 and 3.0
by Alex Wolfson 19 Aug '11

19 Aug '11

> 2011/8/18 Takashi Iwai<tiwai at suse.de <http://mailman.alsa-project.org/mailman/listinfo/alsa-devel>>: > >/ At Tue, 16 Aug 2011 13:31:52 +0100, > />/ Colin Guthrie wrote: > />>/ > />>/ > diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c > />>/ > index c04d7c7..cdd19d7 100644 > />>/ > --- a/sound/usb/mixer.c > />>/ > +++ b/sound/usb/mixer.c > />>/ > @@ -152,6 +152,7 @@ static inline void check_mapped_dB(const struct usbmix_name_map *p, > />>/ > if (p&& p->dB) { > />>/ > cval->dBmin = p->dB->min; > />>/ > cval->dBmax = p->dB->max; > />>/ > + cval->initialized = 1; > />>/ > } > />>/ > } > />>/ > > />>/ > @@ -1092,7 +1093,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, > />>/ > " Switch" : " Volume"); > />>/ > if (control == UAC_FU_VOLUME) { > />>/ > check_mapped_dB(map, cval); > />>/ > - if (cval->dBmin< cval->dBmax) { > />>/ > + if (cval->dBmin< cval->dBmax || !cval->initialized) { > />>/ > kctl->tlv.c = mixer_vol_tlv; > />>/ > kctl->vd[0].access |= > />>/ > SNDRV_CTL_ELEM_ACCESS_TLV_READ | > />>/ > />>/ > />>/ > />>/ Not had a chance to test this yet but I will when I get a moment. > />/ > />/ Any chance to test the patch? > />/ > / > The patch seem fix the no dB scale problem , however the broken dB > test which you added seem redundant as cval->dBmin and cval->dBmax are > still zero because error in get_min_max() at build_feature_ctl() and > it seem that the valid dBmin and dBMax are calculated by get_min_max() > at mixer_ctl_feature_info() on my usb logitech web cam > > > LSA endpoint.c:450: 2:3:1: add audio endpoint 0x86 > ALSA mixer.c:1182: [5] FU [Mic Capture Switch] ch = 1, val = 0/1/1 > ALSA mixer.c:804: 5:2: cannot get min/max values for control 2 (id 5) > ALSA mixer.c:1182: [5] FU [Mic Capture Volume] ch = 1, val = 0/1/1 > ALSA mixer.c:804: 5:2: cannot get min/max values for control 2 (id 5) > usbcore: registered new interface driver snd-usb-audio Fixed dB scale problem for my USB Logitech headset as well (046d:0a0b)

1 0

Re: [alsa-devel] [PATCH] ASoC: sta32x: Move resource allocation and release to the corresponding callback functions
by Mark Brown 18 Aug '11

18 Aug '11

On Thu, Aug 18, 2011 at 03:31:04PM +0800, Axel Lin wrote: > This patch includes below small fixes: > > 1. Move sta32x_set_bias_level() from sta32x_i2c_remove() to sta32x_remove(). > 2. Remove a redundant regulator_bulk_free() call in sta32x_i2c_remove(), > as we will call regulator_bulk_free() in sta32x_remove(). > 3. Remove unneeded snd_soc_codec_set_drvdata(codec, NULL) in sta32x_i2c_remove. > The i2c core will set the clientdata to NULL. Applied, thanks.

1 0

[alsa-devel] [PATCH] ASoC: omap: Fix build errors in ams-delta
by Jarkko Nikula 18 Aug '11

18 Aug '11

Fix "error: too few arguments to function 'ams_delta_set_bias_level'" build errors in ams-delta.c that were introduced after commit d4c6005 ("ASoC: Add context parameter to card DAPM callbacks") by adding dapm context to ams_delta_set_bias_level calls. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Cc: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- sound/soc/omap/ams-delta.c | 6 ++++-- 1 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 30fe0d0..0aa475f 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -514,7 +514,7 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd) } /* Set codec bias level */ - ams_delta_set_bias_level(card, SND_SOC_BIAS_STANDBY); + ams_delta_set_bias_level(card, dapm, SND_SOC_BIAS_STANDBY); /* Add hook switch - can be used to control the codec from userspace * even if line discipline fails */ @@ -649,7 +649,9 @@ static void __exit ams_delta_module_exit(void) ams_delta_hook_switch_gpios); /* Keep modem power on */ - ams_delta_set_bias_level(&ams_delta_audio_card, SND_SOC_BIAS_STANDBY); + ams_delta_set_bias_level(&ams_delta_audio_card, + &ams_delta_audio_card.rtd[0].codec->dapm, + SND_SOC_BIAS_STANDBY); platform_device_unregister(cx20442_platform_device); platform_device_unregister(ams_delta_audio_platform_device); -- 1.7.5.4

3 2

[alsa-devel] [PATCH 1/4] ASoC: DAPM: Allow multiple mixer sources to be routed via the same switch
by Lars-Peter Clausen 18 Aug '11

18 Aug '11

Currently it is only possible to route one source per switch into a mixer. This patch modifies the code, so that it is possible to route multiple sources into a mixer via the same switch. One use-case for this is routing a stereo channel pair into a mono-mixer via the same switch. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-dapm.c | 6 +++++- 1 files changed, 5 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c265311..170c4ff 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -443,6 +443,11 @@ static int dapm_new_mixer(struct snd_soc_dapm_widget *w) if (path->name != (char *)w->kcontrol_news[i].name) continue; + if (w->kcontrols[i]) { + path->kcontrol = w->kcontrols[i]; + continue; + } + wlistsize = sizeof(struct snd_soc_dapm_widget_list) + sizeof(struct snd_soc_dapm_widget *), wlist = kzalloc(wlistsize, GFP_KERNEL); @@ -1556,7 +1561,6 @@ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, /* found, now check type */ found = 1; path->connect = connect; - break; } if (found) -- 1.7.2.5

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Re: [alsa-devel] [PATCH] ASoC: sta32x: Move resource allocation and release to the corresponding callback functions
by Liam Girdwood 18 Aug '11

18 Aug '11

On 18/08/11 08:31, Axel Lin wrote: > This patch includes below small fixes: > > 1. Move sta32x_set_bias_level() from sta32x_i2c_remove() to sta32x_remove(). > 2. Remove a redundant regulator_bulk_free() call in sta32x_i2c_remove(), > as we will call regulator_bulk_free() in sta32x_remove(). > 3. Remove unneeded snd_soc_codec_set_drvdata(codec, NULL) in sta32x_i2c_remove. > The i2c core will set the clientdata to NULL. > > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> Acked-by: Liam Girdwood <lrg(a)ti.com>

1 0

Re: [alsa-devel] [PATCH] ASoC: sta32x: Move resource allocation and release to the corresponding callback functions
by Johannes Stezenbach 18 Aug '11

18 Aug '11

On Thu, Aug 18, 2011 at 03:31:04PM +0800, Axel Lin wrote: > This patch includes below small fixes: > > 1. Move sta32x_set_bias_level() from sta32x_i2c_remove() to sta32x_remove(). > 2. Remove a redundant regulator_bulk_free() call in sta32x_i2c_remove(), > as we will call regulator_bulk_free() in sta32x_remove(). > 3. Remove unneeded snd_soc_codec_set_drvdata(codec, NULL) in sta32x_i2c_remove. > The i2c core will set the clientdata to NULL. > > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> > --- > hi, > I don't have this hardware handy. I appreciate if someone can test this patch. > > Axel Your patch looks good, but I also don't have the hardware currently. Maybe Sven or Daniel could test? But even though it's untested: Acked-by: Johannes Stezenbach <js(a)sig21.net> Thanks Johannes > sound/soc/codecs/sta32x.c | 13 ++----------- > 1 files changed, 2 insertions(+), 11 deletions(-) > > diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c > index 3d155f5..5c7def3 100644 > --- a/sound/soc/codecs/sta32x.c > +++ b/sound/soc/codecs/sta32x.c > @@ -812,6 +812,7 @@ static int sta32x_remove(struct snd_soc_codec *codec) > { > struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); > > + sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF); > regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); > regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); > > @@ -871,18 +872,8 @@ static __devinit int sta32x_i2c_probe(struct i2c_client *i2c, > static __devexit int sta32x_i2c_remove(struct i2c_client *client) > { > struct sta32x_priv *sta32x = i2c_get_clientdata(client); > - struct snd_soc_codec *codec = sta32x->codec; > - > - if (codec) > - sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF); > - > - regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); > - > - if (codec) { > - snd_soc_unregister_codec(&client->dev); > - snd_soc_codec_set_drvdata(codec, NULL); > - } > > + snd_soc_unregister_codec(&client->dev); > kfree(sta32x); > return 0; > } > -- > 1.7.4.1 > > > >

1 0

Re: [alsa-devel] Correct way to set NONAUDIO flag
by Clemens Ladisch 17 Aug '11

17 Aug '11

Pierre-Louis Bossart wrote: > If you try to set the bits with ALSA controls in PulseAudio, somehow the > settings are overwritten when the iec958 device is opened and initialized > with default values. That's how the iec958 device is intended to work. The AES0..3 parameters of the iec958 device are the _only_ way to set these flags. If you don't set them, they get a static default value; see <http://git.alsa-project.org/?p=alsa-lib.git;a=blob;hb=HEAD;f=src/conf/pcm/i…>. If PA wants to set these flags, it must do so when opening the device; any other mechanism will not work. Regards, Clemens

1 0

Re: [alsa-devel] Correct way to set NONAUDIO flag
by Arun Raghavan 17 Aug '11

17 Aug '11

On Wed, 2011-08-17 at 10:31 -0500, Pierre-Louis Bossart wrote: > > > > I'm trying to figure out the right way to set the NONAUDIO flag on > > an > > > > iec958 device. > > > > > > Use the device name "iec958:AES0=6". > > > > Thanks. As I understand it, this forces us to set all the flags instead > > of just the one that we're interested in. Is this the only way? > > I banged my head against the wall on this one for eAC3 support. For some > strange reason my receiver works fine with AC3 without touching the C bits, > but for eAC3 setting the NONAUDIO bit is required. > If you try to set the bits with ALSA controls in PulseAudio, somehow the > settings are overwritten when the iec958 device is opened and initialized > with default values. Clemens is right, setting the AES0=6 in the hdmi > profile proved to be the only solution. It'd be nice to find a solution in > which the iec958 device preserves the AES values set beforehand. That way we > could set the NONAUDIO bit only in passthrough mode. Well, given this is the best way to do it, my plan is to just close (suspend) and reopen (unsuspend) and tack on the the right flags to the device name when we're in passthrough mode. -- Arun

1 0

[alsa-devel] Correct way to set NONAUDIO flag
by Arun Raghavan 17 Aug '11

17 Aug '11

Hello, I'm trying to figure out the right way to set the NONAUDIO flag on an iec958 device. The idea is that when we get a new stream in PulseAudio that signals a non-PCM format, we'd like to set the NONAUDIO flag to let receivers know that this isn't PCM data. I tried doing what iecset does in the PA alsa-sink code but this fails. As far as I can tell, this happens because: (a) if I set the flag before snd_pcm_open(), the AES0 value from iec958.conf gets used, and (b) if I set the flag after snd_pcm_open(), changing the value is no longer permitted. Any help/pointers would be appreciated. Thanks, Arun

3 5

[PATCH 1/1] sgtl5000: fix record unwork issue
by Dong Aisheng 17 Aug '11

17 Aug '11

Signed-off-by: Dong Aisheng <b29396(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 32 ++++++++++++++++++-------------- 1 files changed, 18 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 5a0d8e4..ef52890 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -198,13 +198,13 @@ static int small_pop_event(struct snd_soc_dapm_widget= *w, { switch (event) { case SND_SOC_DAPM_PRE_PMU: - snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, - SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP); +// snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, +// SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP); break; =20 case SND_SOC_DAPM_PRE_PMD: - snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, - SGTL5000_VAG_POWERUP, 0); +// snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, +// SGTL5000_VAG_POWERUP, 0); msleep(400); break; default: @@ -1153,11 +1153,13 @@ static int sgtl5000_set_power_regs(struct snd_soc_c= odec *codec) SGTL5000_VDDC_CHRGPMP_POWERUP, 0); =20 /* VDDC use VDDIO rail */ - lreg_ctrl |=3D SGTL5000_VDDC_ASSN_OVRD; - lreg_ctrl |=3D SGTL5000_VDDC_MAN_ASSN_VDDIO << - SGTL5000_VDDC_MAN_ASSN_SHIFT; +// lreg_ctrl |=3D SGTL5000_VDDC_ASSN_OVRD; +// lreg_ctrl |=3D SGTL5000_VDDC_MAN_ASSN_VDDIO << +// SGTL5000_VDDC_MAN_ASSN_SHIFT; } =20 +// ana_pwr |=3D SGTL5000_VDDC_CHRGPMP_POWERUP | SGTL5000_VAG_POWERUP | SGT= L5000_HP_POWERUP; + ana_pwr |=3D 0x5afb; snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL, lreg_ctrl); =20 snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, ana_pwr); @@ -1194,9 +1196,9 @@ static int sgtl5000_set_power_regs(struct snd_soc_cod= ec *codec) else vag =3D (vag - SGTL5000_ANA_GND_BASE) / SGTL5000_ANA_GND_STP; =20 - snd_soc_update_bits(codec, SGTL5000_CHIP_REF_CTRL, - vag << SGTL5000_ANA_GND_SHIFT, - vag << SGTL5000_ANA_GND_SHIFT); +// snd_soc_update_bits(codec, SGTL5000_CHIP_REF_CTRL, +// vag << SGTL5000_ANA_GND_SHIFT, +/// vag << SGTL5000_ANA_GND_SHIFT); =20 /* set line out VAG to vddio / 2, in range (0.8v, 1.675v) */ vag =3D vddio / 2; @@ -1208,14 +1210,15 @@ static int sgtl5000_set_power_regs(struct snd_soc_c= odec *codec) else vag =3D (vag - SGTL5000_LINE_OUT_GND_BASE) / SGTL5000_LINE_OUT_GND_STP; - +#if 1 snd_soc_update_bits(codec, SGTL5000_CHIP_LINE_OUT_CTRL, - vag << SGTL5000_LINE_OUT_GND_SHIFT | +// vag << SGTL5000_LINE_OUT_GND_SHIFT | SGTL5000_LINE_OUT_CURRENT_360u << SGTL5000_LINE_OUT_CURRENT_SHIFT, - vag << SGTL5000_LINE_OUT_GND_SHIFT | +// vag << SGTL5000_LINE_OUT_GND_SHIFT | SGTL5000_LINE_OUT_CURRENT_360u << SGTL5000_LINE_OUT_CURRENT_SHIFT); +#endif =20 return 0; } @@ -1378,7 +1381,8 @@ static int sgtl5000_probe(struct snd_soc_codec *codec= ) =20 snd_soc_write(codec, SGTL5000_CHIP_ANA_CTRL, SGTL5000_HP_ZCD_EN | - SGTL5000_ADC_ZCD_EN); + SGTL5000_ADC_ZCD_EN=20 + | SGTL5000_LINE_OUT_MUTE); =20 snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 0); =20 --=20 1.7.0.4

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[alsa-devel] How to pair Wine with ALSA? part1: intro & underruns (long)
by Joerg-Cyril.Hoehle＠t-systems.com 17 Aug '11

17 Aug '11

Dear knowledgeable ALSA developers, Wine currently undergoes a rewrite of its audio systems. That's a welcome point in time to reflect on what it needs from ALSA and how it should use the ALSA API. What combination of hw & sw_params makes sense? How to feed ALSA with samples? I've identified 3 uses cases and several other topics to help thinking about the issues: UC1 Intermittent Mouse Clicks UC2 Video UC3 Background Music T1 underrun T2 buffer and period size/time T3 blocking or not T4 mmap Due to size, I'll cover T2-T4 in a later e-mail. UC1 Intermittent Mouse Clicks An app may use the winmm API as follows: 1. open the device 2. from time to time, e.g. on mouse clicks, send wave data. The requirements are: R1 Nothing (or silence) is to be played when the app submits no data. R2 Once the app submits data, it should play ASAP. That is the app's view on it. It seems to translate to silence_threshold=0 and silence_size=boundary, doesn't it? Alternatively, Wine's internal periodic audio worker could as well send silence samples when it receives nothing from the app. If so, I fear additional complexity with snd_pcm_rewind to meet R2, e.g. in case where the app submits samples shortly after wine decided to play some silence. UC2 Video I've not much to say here because I basically ignore how apps synchronize audio with video. GetPosition returns "the stream position of the sample that is currently playing through the speakers". This begs for snd_pcm_delay, not avail_update. An app may select a particular device for output (e.g. 5:1 may not be available with the default mixer). R3 Wine must offer access to several sound cards if available. R4 Try and find hw/sw_params that work with the "default" device as well as individual cards (hw:x or perhaps plughw:x). mmdevapi has a notion of exclusive and shared modes. I expect shared mode to be asked with the "default" device (for mixing to work), and exclusive with others (e.g. 5:1 sound). OTOH, apps may also ask for the default device in exclusive mode, which is said to be granted if no other app is playing sound (much like I can grab hw:0 only when dmix has nothing to do). UC3 Background Music The app ought to send a continuous stream of samples, perhaps mixing sources itself (e.g. what games let the dsound API do). R5 No samples are ever skipped. If late, play late. See R1/R2. I've observed that "dmix"+"pulse" and the "hw:x" devices approach underruns radically differently. In case of a 10s underrun, the former silently skip over the next 10s worth of samples(!) -- if the app does not submit those *fast* to catch up, you'll hear no more sound! -- while the latter immediately write them to their HW buffer. IOW, dmix and pulse streaming semantics do not match winmm nor mmdevapi's semantics. To counter that, I'm considering using: /* call avail_update prior to every write: */ if (snd_pcm_avail_update() > buffer_size) snd_pcm_reset() /* skip over late samples */ /* should be equivalent to snd_pcm_forward(avail) */ snd_pcm_write() However, there's still a short underrun between reset() and write(). Topic T1 underrun I argue that the MS APIs winmm and mmdevapi have almost no notion of underrun during playback. You derive it indirectly at best ("buffer is empty", "all headers returned", "speaker position = amount of samples sent"). R5 GetPosition only counts submitted samples. Silence played during underruns must not count and is not reported to the app. This seems unlike snd_pcm_delay. I've observed snd_pcm_delay grow towards minus infinity during underruns. How to nevertheless make use of the ALSA API to compute that? I'm thinking about err = snd_pcm_avail_delay(&avail,&delay) if (err<0) ...? if (avail > buffer_size) /* underrun => everything was played */ position = sum_of_submitted_samples; else position = some_function_of(delay, submitted_samples) What do you recommend? Thank you for your help and for reading up to this point, Jörg Höhle

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Re: [alsa-devel] snd soc spi read/write
by Scott Jiang 17 Aug '11

17 Aug '11

> > Why are you replying off-list? You should implement 16x8 SPI reads in > the core stuff I'd expect, then disable cache for the CODEC. > Sorry, I just forgot to CC. I wonder if rbtree compress type can support this kind of sparse register? And should the patch aginst 3.0 or 3.2?

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Re: [alsa-devel] [PATCH] ASoC: Remove unreachable code in au1xac97c_drvprobe and au1xi2s_drvprobe
by Mark Brown 17 Aug '11

17 Aug '11

On Wed, Aug 17, 2011 at 03:14:33PM +0800, Axel Lin wrote: > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> Applied, thanks.

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[alsa-devel] [PATCH] ASoC: Fix check for symmetric rate enforcement
by Sascha Hauer 17 Aug '11

17 Aug '11

The ASoC core tries to not enforce symmetric rates when two streams open simultaneously. It does so by checking rtd->rate being zero. This works exactly once after booting because it is not set to zero again when the streams close. Fix this by clearing rtd->rate when no active stream is left. Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> --- sound/soc/soc-core.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d75043e..c25649a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -746,6 +746,9 @@ static int soc_codec_close(struct snd_pcm_substream *substream) codec_dai->active--; codec->active--; + if (!cpu_dai->active && !codec_dai->active) + rtd->rate = 0; + /* Muting the DAC suppresses artifacts caused during digital * shutdown, for example from stopping clocks. */ -- 1.7.5.4 -- Pengutronix e.K. | | Industrial Linux Solutions | http://www.pengutronix.de/ | Peiner Str. 6-8, 31137 Hildesheim, Germany | Phone: +49-5121-206917-0 | Amtsgericht Hildesheim, HRA 2686 | Fax: +49-5121-206917-5555 |

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[alsa-devel] [PATCH] ASoC: fsl: fix build warning in fsl_dma
by Timur Tabi 17 Aug '11

17 Aug '11

The previous patch to fsl_dma.c ("fix initialization of DMA buffers") left behind an unused local variable that causes a build warning. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/fsl_dma.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 732208c..0efc04a 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -297,7 +297,6 @@ static irqreturn_t fsl_dma_isr(int irq, void *dev_id) static int fsl_dma_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; static u64 fsl_dma_dmamask = DMA_BIT_MASK(36); int ret; -- 1.7.4.4

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Re: [alsa-devel] USB Audio questions
by Torsten Schenk 17 Aug '11

17 Aug '11

Hi Pierre, I also spent some time learning about USB and audio while developing the sound/usb/6fire driver. I made the following discoveries: >What I had in mind is a case where the ring buffer is divided in separate >periods, which would each correspond to an integer number of URBs/packets. >This way there is _never_ a case of wrap-around and no need for memcpys. >It's probably less generic than the current driver but a whole lot >simpler/efficient. I was also not very happy to implement the double buffering with memcpy's. But from the experience with the 6fire device I can say this: - The device/driver sends per urb a constant number of packets (as you already also discovered). The number of samples per packet differ however (+- 1 sample per packet). I accuse the asynchronousity between the urb interval and the sound card's internal clock for this fact. The device expects the driver to send back an urb with exactly the same structure. The device produced crackling noise if the samples per packet were not equal. I cannot say to what extend this is true (i.e. if you could wait some packets until catching up the difference without producing crackles) but due to this fact it's at least not a simple task to send urbs exactly corresponding to the period limit. - Another thing came into my mind while reading your post, but I'm not sure if that would apply in praxis: If sending an urb is scheduled with the alsa buffer, it might be possible that the buffer is overwritten before the urb is sent, leading to complete audio-rubbish. - And: at least the 6fire requires a header for each packet, which couldn't be placed directly in the alsa buffer. But perhaps this doesn't apply to most cards. Greets, Torsten

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[alsa-devel] WM8727 & AT91SAM9G45: not enough memory
by conn intel 17 Aug '11

17 Aug '11

Dear Friends, I am using WM8727 codec with AT91SAM9G45 processor. I have implemented machine code as http://pastebin.com/axadqGhd . After insterting the module I got following output: # insmod sam9g45_wm8727.ko atmel-pcm:preallocate_dma_buffer: area=ffadf000, addr=77c40000, size=32768 asoc: WM8727 <-> atmel-ssc0 mapping ok I am getting following error when I tried to play the wav file using command "aplay -D default sample_16.wav ": atmel_ssc_startup: SSC_SR=0x192 Playing WAVE 'sample_16.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo atmel_ssc_hw_params: RCMR=00010721 RFMR=0000008f TCMR=00010702 TFMR=0000008f atmel_ssc_dai: Starting clock atmel_ssc_dai,hw_params: SSC initialized atmel-pcm: hw_params: DMA for SSC0 PCM out initialized (dma_bytes=32768, period_size=8192) transmit enabled SSC_SR=0x000104c3 aplayatmel_ssc_shutdown: transmit disabled SSC_SR=0x000000c3 :atmel_ssc_dau: Stopping clock set_params:1167: not enough memory Does anyone throw some light where and how to solve the issue? Kindly share if one needs further information. Have a Good Day, Thank you, Ankur.

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[alsa-devel] [PATCH] ASoC: Fix backport of WM8994 thermal warning
by Mark Brown 17 Aug '11

17 Aug '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> --- sound/soc/codecs/wm8994.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 9412491..e537267 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3141,9 +3141,9 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994_request_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, wm8994_fifo_error, "FIFO error", codec); - wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_TEMP_WARN, + wm8994_request_irq(wm8994->control_data, WM8994_IRQ_TEMP_WARN, wm8994_temp_warn, "Thermal warning", codec); - wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_TEMP_SHUT, + wm8994_request_irq(wm8994->control_data, WM8994_IRQ_TEMP_SHUT, wm8994_temp_shut, "Thermal shutdown", codec); ret = wm8994_request_irq(codec->control_data, WM8994_IRQ_DCS_DONE, -- 1.7.5.4

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[alsa-devel] Playback PCM drain and card fifos
by Eliot Blennerhassett 17 Aug '11

17 Aug '11

Greetings, I am seeing playback audio get truncated at the end of playback. The amount lost is about what is in the on-card fifo. (asihpi driver) For example in a specific test this is about 100ms of audio, though in theory could go much higher if I let the card fifo fill completely. The current driver calls snd_pcm_period_elapsed when the card has read from the alsa ringbuffer. I.e. it is notifying that the buffer is available for more data to be written. As soon as the driver notifies that the final data has been read by the card, the trigger stop is called, which in turn stops the card playback immediately. However, the data just read has not been played back yet so the audio is truncated. During playback the driver sets runtime->delay = on_card_bytes I'm trying to work out what I can do in the driver (or in alsa mid layer if necessary), to make this work better? regards -- Eliot Blennerhassett AudioScience Inc.

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[alsa-devel] [PATCH] [v2] ASoC: claim the IRQ when the fsl_ssi device is probed, not opened
by Timur Tabi 17 Aug '11

17 Aug '11

The PowerPC Freescale SSI driver is claiming the IRQ when the IRQ when the device is opened, which means that the /proc/interrupts entry for the SSI exists only during playback or capture. This also meant that the user won't know that the IRQ number is wrong until he tries to use the device. Instead, we should claim the IRQ when the device is probed. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 61 ++++++++++++++++++++++++++++------------------ 1 files changed, 37 insertions(+), 24 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index d48afea..06ac2b9 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -289,16 +289,6 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, */ if (!ssi_private->playback && !ssi_private->capture) { struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - int ret; - - /* The 'name' should not have any slashes in it. */ - ret = request_irq(ssi_private->irq, fsl_ssi_isr, 0, - ssi_private->name, ssi_private); - if (ret < 0) { - dev_err(substream->pcm->card->dev, - "could not claim irq %u\n", ssi_private->irq); - return ret; - } /* * Section 16.5 of the MPC8610 reference manual says that the @@ -522,15 +512,12 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, ssi_private->second_stream = NULL; /* - * If this is the last active substream, disable the SSI and release - * the IRQ. + * If this is the last active substream, disable the SSI. */ if (!ssi_private->playback && !ssi_private->capture) { struct ccsr_ssi __iomem *ssi = ssi_private->ssi; clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); - - free_irq(ssi_private->irq, ssi_private); } } @@ -675,17 +662,30 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) ret = of_address_to_resource(np, 0, &res); if (ret) { dev_err(&pdev->dev, "could not determine device resources\n"); - kfree(ssi_private); - return ret; + goto error_kmalloc; } ssi_private->ssi = of_iomap(np, 0); if (!ssi_private->ssi) { dev_err(&pdev->dev, "could not map device resources\n"); - kfree(ssi_private); - return -ENOMEM; + ret = -ENOMEM; + goto error_kmalloc; } ssi_private->ssi_phys = res.start; + ssi_private->irq = irq_of_parse_and_map(np, 0); + if (ssi_private->irq == NO_IRQ) { + dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); + ret = -ENXIO; + goto error_iomap; + } + + /* The 'name' should not have any slashes in it. */ + ret = request_irq(ssi_private->irq, fsl_ssi_isr, 0, ssi_private->name, + ssi_private); + if (ret < 0) { + dev_err(&pdev->dev, "could not claim irq %u\n", ssi_private->irq); + goto error_irqmap; + } /* Are the RX and the TX clocks locked? */ if (of_find_property(np, "fsl,ssi-asynchronous", NULL)) @@ -711,7 +711,7 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) if (ret) { dev_err(&pdev->dev, "could not create sysfs %s file\n", ssi_private->dev_attr.attr.name); - goto error; + goto error_irq; } /* Register with ASoC */ @@ -720,7 +720,7 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) ret = snd_soc_register_dai(&pdev->dev, &ssi_private->cpu_dai_drv); if (ret) { dev_err(&pdev->dev, "failed to register DAI: %d\n", ret); - goto error; + goto error_dev; } /* Trigger the machine driver's probe function. The platform driver @@ -741,18 +741,28 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) if (IS_ERR(ssi_private->pdev)) { ret = PTR_ERR(ssi_private->pdev); dev_err(&pdev->dev, "failed to register platform: %d\n", ret); - goto error; + goto error_dai; } return 0; -error: +error_dai: snd_soc_unregister_dai(&pdev->dev); + +error_dev: dev_set_drvdata(&pdev->dev, NULL); - if (dev_attr) - device_remove_file(&pdev->dev, dev_attr); + device_remove_file(&pdev->dev, dev_attr); + +error_irq: + free_irq(ssi_private->irq, ssi_private); + +error_irqmap: irq_dispose_mapping(ssi_private->irq); + +error_iomap: iounmap(ssi_private->ssi); + +error_kmalloc: kfree(ssi_private); return ret; @@ -766,6 +776,9 @@ static int fsl_ssi_remove(struct platform_device *pdev) snd_soc_unregister_dai(&pdev->dev); device_remove_file(&pdev->dev, &ssi_private->dev_attr); + free_irq(ssi_private->irq, ssi_private); + irq_dispose_mapping(ssi_private->irq); + kfree(ssi_private); dev_set_drvdata(&pdev->dev, NULL); -- 1.7.4.4

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[alsa-devel] [PATCH] ASoC: claim the IRQ when the fsl_ssi device is probed, not opened
by Timur Tabi 17 Aug '11

17 Aug '11

The PowerPC Freescale SSI driver is claiming the IRQ when the IRQ when the device is opened, which means that the /proc/interrupts entry for the SSI exists only during playback or capture. This also meant that the user won't know that the IRQ number is wrong until he tries to use the device. Instead, we should claim the IRQ when the device is probed. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- arch/powerpc/configs/mpc85xx_defconfig | 1 + arch/powerpc/configs/mpc85xx_smp_defconfig | 1 + 2 files changed, 2 insertions(+), 0 deletions(-) diff --git a/arch/powerpc/configs/mpc85xx_defconfig b/arch/powerpc/configs/mpc85xx_defconfig index fcd85d2..a3467bf 100644 --- a/arch/powerpc/configs/mpc85xx_defconfig +++ b/arch/powerpc/configs/mpc85xx_defconfig @@ -139,6 +139,7 @@ CONFIG_SND=y CONFIG_SND_INTEL8X0=y # CONFIG_SND_PPC is not set # CONFIG_SND_USB is not set +CONFIG_SND_SOC=y CONFIG_HID_A4TECH=y CONFIG_HID_APPLE=y CONFIG_HID_BELKIN=y diff --git a/arch/powerpc/configs/mpc85xx_smp_defconfig b/arch/powerpc/configs/mpc85xx_smp_defconfig index 908c941..9693f6e 100644 --- a/arch/powerpc/configs/mpc85xx_smp_defconfig +++ b/arch/powerpc/configs/mpc85xx_smp_defconfig @@ -140,6 +140,7 @@ CONFIG_SND=y CONFIG_SND_INTEL8X0=y # CONFIG_SND_PPC is not set # CONFIG_SND_USB is not set +CONFIG_SND_SOC=y CONFIG_HID_A4TECH=y CONFIG_HID_APPLE=y CONFIG_HID_BELKIN=y -- 1.7.4.4

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Re: [alsa-devel] USB Audio questions
by Torsten Schenk 17 Aug '11

17 Aug '11

> Thanks for sharing your experience. > > > - The device/driver sends per urb a constant number of packets (as you > > already also discovered). The number of samples per packet differ > > however (+- 1 sample per packet). I accuse the asynchronousity between > > the urb interval and the sound card's internal clock for this fact. The > > device expects the driver to send back an urb with exactly the same > > structure. The device produced crackling noise if the samples per > > packet were not equal. I cannot say to what extend this is true (i.e. > > if you could wait some packets until catching up the difference without > > producing crackles) but due to this fact it's at least not a simple > > task to send urbs exactly corresponding to the period limit. > > I am still unclear on what makes the number of samples per packet vary, I > didn't see this in the code. I understand that it could vary for 44.1 but > for 48kHz (my case) I'd think it should be constant. The variation is not directly in the code. I remember that the caiaq driver enlightened me since this behaviour is commented there. The variation comes from the device and these steps lead to this variation: 1. the driver sends n urbs containing m packets with size 0 (i.e. no audio) to the device 2. the device sends back each of the n urbs one-by-one, filled with the captured audio; the urbs will contain m packets, each packet having an individual number of samples s_p ("sample" being defined here as: 1 sample contains one value for each channel) 3. for each urb returned from the device, the driver sends another urb containing again m packets, but this time with s_p samples per packet (corresponding to the packet-dependent s_p received from the device) I indeed found out that when using 48kHz, I first thought that the number of samples per packet were identical every time. When using this heuristic, crackling noise occured in the sound output and I looked at the number of samples per packet. They varied not very often but sometimes - around once per 2 seconds. My explanation for this problem is, that the sound card has its own clock, that will not oscillate at exactly an integer multiple as the computer's/usb's internal clock. Even temperature will change the frequency a little bit, meaning that 48kHz is not always exactly 48kHz. I would have to start speculating about which clock drives the usb urb scheduling, so this question should be left for the more experienced USB programmers. > > > - Another thing came into my mind while reading your post, but I'm not > > sure if that would apply in praxis: If sending an urb is scheduled with > > the alsa buffer, it might be possible that the buffer is overwritten > > before the urb is sent, leading to complete audio-rubbish. > > Indeed there could be some issues with pulseaudio rewinds, but the fact is > that rewinds are broken anyway when the data is copied... I gave this another thought and came to the conclusion: if you'd call snd_pcm_period_elapsed in the output urb retire callback, the theoretical problem wouldn't occur. Torsten

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[alsa-devel] [HELP WANTED] ALSA Driver developer
by Gabriel Beddingfield 16 Aug '11

16 Aug '11

All: A recruiter approached me looking for an ALSA driver developer to enhance an existing driver. This is a short-term, off-site gig. Some details below. Please contact Mr. DeAmelio directly. Thanks, Gabriel ---------- Forwarded message ---------- From: David DeAmelio <david_deamelio(a)oxfordcorp.com> Date: Tue, Aug 16, 2011 at 2:29 PM Subject: Job Description - Oxford International To: gabrbedd(a)gmail.com Hey Gabriel, Here’s the description….Thanks. Location: OFFSITE Start Date: ASAP Duration: 1-2 Weeks Manager needs consultant to come onsite for 1st few days. Client will pay for flight there and back. Must: Linux ALSA Device Drivers—Alsa- advanced linux sound architecture Blackfin Plus: D2D: Consultant will be enhancing an audio driver to support a voice quality recorder __________________________________ David De Amelio Team Lead Software/Hardware Consulting Services Oxford International a division of Oxford Global Resources 100 Cummings Center Suite 206N Beverly, MA 01915 877.258.9982 x 7608 Office 978.922.7547 FAX david_deamelio(a)oxfordcorp.com www.oxfordcorp.com Oxford Europe The Right Talent. Right Now.® This e-mail, and any attachments thereto, is intended only for the addressee(s) named herein and may contain privileged and/or confidential information. If you are not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail, and any attachments thereto, is strictly prohibited. If you have received this e-mail in error, immediately notify the sender by return e-mail and permanently delete the original, including any attachments thereto, and any copy and printout thereof.

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Re: [alsa-devel] USB Audio questions
by Clemens Ladisch 16 Aug '11

16 Aug '11

Pierre-Louis Bossart wrote: > What is this USB frame counter and what would to enable its use, at > least for playback where there is scope for lots of power > optimizations. USB timing is defined by frames, of which there are about 1000 per second. The audio driver schedules one packet to be sent/received for each frame. Multiple packets are grouped in a URB (USB request buffer); the controller raises an interrupt when the last packet of the URB has been completed. By reading the current frame number, the driver could deduce how many packets (and therefore, how many samples) have been processed by the controller. > > Why more efficient? You cannot avoid double-buffering when a USB packet > > wraps around the end of the ALSA ring buffer. > > What I had in mind is a case where the ring buffer is divided in separate > periods, which would each correspond to an integer number of URBs/packets. This works only when the driver has control over how many samples actually are transmitted in each USB frame. The playback streams of many USB audio devices work this way, but quite a few others, and capture streams, don't. > It's probably less generic than the current driver but a whole lot > simpler/efficient. The code wouldn't be simpler because we'd still need the other algorithm. This would be just an optimization. Regards, Clemens

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Re: [alsa-devel] USB Audio questions
by Clemens Ladisch 16 Aug '11

16 Aug '11

Pierre-Louis Bossart wrote: > - Is there any good reason why the max number of packets per urb defaults to > 8? Not really. It is an attempt to compromise between interrupt frequency and the latency resulting from packet queueing. > - Increasing the number of packets/urbs solves my power issue but not the > synchronization issue. If I reduce the number of urbs to reduce the > interrupt rate, then the accuracy of the hw_pointer is decreased big time > and it becomes difficult to synchronize with video. The .pointer routine > only keeps track of retired urbs, it seems there's no way to know how many > samples were played at a given time? Only the URB completion callback tells us about finished packets. For playback, we know how many bytes we had submitted, so we could derive the current position from the USB frame counter, but for capture, we don't know the precise byte count until the HCD has looked at the descriptors. > Isn't there a direct way to query the usb core to know how many bytes > have been transmitted and make the hw_ptr/delay information independent > of the urb buffer size? No, this information isn't returned earlier than the completion callback. It would be nice to be able to tell the HCD to collect information about all currently completed USB frames, regardless of whether the entire URB has finished or whether its interrupt flag is set, but this would require changes in all host controller drivers. (I'm contemplating something like this for FireWire, but there we have only one controller driver.) > - Is there any reason why the driver relies on vmalloc and memcpy? It seems > like using physically-contiguous memory would be more efficient, as done in > the CAIQ driver? Why more efficient? You cannot avoid double-buffering when a USB packet wraps around the end of the ALSA ring buffer. > - The period information is not updated when the urbs are allocated. In my > case the period specified was 25ms, but with all the approximations the > actual period (number of urbs*buffer_size_per_urb) was 24ms. Please note that URBs are usually not completely filled. For playback, the driver makes URBs shorter so that they end as nearly as possible on or after a period boundary; for capture, we cannot predict sample counts, so the driver just uses one URB per ms. Regards, Clemens

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Re: [alsa-devel] USB Audio questions
by Pierre-Louis Bossart 16 Aug '11

16 Aug '11

> For USB 3.0 host controllers, the HCD doesn't get the information about > which frame exactly the transfer went out in. This is because all the > scheduling is done in hardware, not software. The xHCI driver can only > record the frame that was current when the interrupt for the URB > happened. > > Pierre, I'm not sure you can get the precise measurements you want out > of a USB 3.0 host controller to do the audio/video sync. You're > welcome > to suggest something to Steve McGowan to add to the xHCI spec, and you > should probably double check with him that what I'm saying is correct. Thanks Sarah for the heads-up. I was only looking at existing USB Audio solutions for now, looks like there's some work to be done for USB3 as well. We do need precise timing information for all audio drivers. Probably a good topic for LPC discussions... -Pierre

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[alsa-devel] [PATCH] ASoC: Run Speyside WM8962 at 512fs
by Mark Brown 16 Aug '11

16 Aug '11

Ensure we have access to all the advanced DSP functinality offered by the WM8962 by running the system clock at 512fs. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/speyside_wm8962.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c index 0b9eb5f..753e1c2 100644 --- a/sound/soc/samsung/speyside_wm8962.c +++ b/sound/soc/samsung/speyside_wm8962.c @@ -28,13 +28,13 @@ static int speyside_wm8962_set_bias_level(struct snd_soc_card *card, if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, WM8962_FLL_MCLK, 32768, - 44100 * 256); + 44100 * 512); if (ret < 0) pr_err("Failed to start FLL: %d\n", ret); ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_FLL, - 44100 * 256, + 44100 * 512, SND_SOC_CLOCK_IN); if (ret < 0) { pr_err("Failed to set SYSCLK: %d\n", ret); -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: Clear completions from late WM8996 FLL lock IRQs
by Mark Brown 16 Aug '11

16 Aug '11

In case we have a pending completion, for example due to a problem with the input clock which got corrected after we timed out. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index ab8e9d1..a44a0e5 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2106,6 +2106,9 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source, snd_soc_write(codec, WM8996_FLL_EFS_1, fll_div.lambda); + /* Clear any pending completions (eg, from failed startups) */ + try_wait_for_completion(&wm8996->fll_lock); + snd_soc_update_bits(codec, WM8996_FLL_CONTROL_1, WM8996_FLL_ENA, WM8996_FLL_ENA); -- 1.7.5.4

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[alsa-devel] How to pair Wine with ALSA? part2: buffer&period size / blocking / mmap (long)
by Joerg-Cyril.Hoehle＠t-systems.com 16 Aug '11

16 Aug '11

Dear ALSA developers, [Please refer to part1:intro & underruns for the introductory text] http://mailman.alsa-project.org/pipermail/alsa-devel/2011-August/042746.html Topic T2 period and buffer size and time Wine apps using the old winmm API cannot tell at waveOutOpen time "give me a large buffer" or "give me fast reaction to explosions". R6 A good compromise is needed for when Wine opens ALSA on behalf of winmm. Years later, the mmdevapi introduced with Vista provides "period" and duration parameters. It also implements sort of a dmix device: it appears to mix at a fixed rate (48000 or 44100 samples/sec, user settable) and to mix data in packets of 10ms. Incidentally, that's why MS claims 10ms latency. For compatibility, Wine should match that rate -- at least when accessing the "default" device. Does that really translate to set_period_time? I doubt it. I wonder why Wine up to now insists on particular ALSA buffer and period sizes. I tend to consider it's none of its business. It might make sense for Wine to use a periodic timer that calls snd_pcm_write (see T3 below). *That* timer should be set to mmdevapi's 10ms. Doesn't that translate to set_period_time_*max*(10ms)? However if ALSA wants/requires 50ms, why not? Furthermore, if the app wants to use large buffers, why insist on a tiny period on the ALSA side? (MS appears to stick to 10ms packets regardless of the app's requested buffer size). I expect that setting Wine's timer period to at least ALSA's allows it to actually find room for new data each turn. So here's what makes sense to me: if (shared_mode && device=="default") set_period_time_near(10ms); else if (exclusive_mode) /* mmdevapi:Initialize receives period from the app */ set_period_time_near(clamp(3ms, app's wish in mmdevapi:Intialize, 100ms)); /* 100ms is arbitrary, why not 1s? */ snd_pcm_hw_params() wine_set_timer_rate(clamp(3ms, snd_pcm_get_period_time(), 0.5s)) Topic T2.b Duration / buffer size mmdevapi's Initialize method receives a duration parameter as a hint towards either small latency or large buffering. One would think that it makes perfect sense to forward that to snd_pcm_set_buffer_time. *However*, mmdevapi also requires to hand out a pointer to a buffer that large (GetBuffer). Thus Wine must maintain a buffer that large (possibly even two of them, for reasons not relevant here). Now should ALSA really keep yet another buffer that large? Isn't that precisely why people have gripes with PA's 2s buffer? I'm wondering whether Wine should solely rely on its periodic timer to regularly submit data and e.g. ask ALSA to use a buffer 3 times the period? 30ms (3x10ms) seems to play a role in MS systems. Dmix seems to prefer 4 to 6 times period size. Unfortunately, snd_pcm_get_period_time may be known only after invoking snd_pcm_hw_params(), so I can't express: set_period_time_near(10ms); set_buffer_time_near(3 x actual_period); Prefer set_buffer_time_near(3 x period_above); or simply not call set_buffer at all? Finally, there's that snd_pcm_sw_params_set_avail_min whose purpose I cannot figure out. Should Wine call snd_pcm_sw_params_set_avail_min(1); or snd_pcm_sw_params_set_avail_min(0); or not at all? Topic T3 blocking or not Wine has traditionally used ALSA in non-blocking mode, which ALSA people recommended against (still?). Now suppose every write is preceded with avail_update, for reasons I gave in part1: underruns. Remember my example from part 1, slightly refined: if (snd_pcm_avail_update(&avail) > buffer_size) snd_pcm_reset() /* skip over late samples */ /* should be equivalent to snd_pcm_forward(avail) */ written = snd_pcm_write(min(avail,frames)); My understanding of the semantics is that write(<avail) will not block, thus Wine could as well dispose of SND_PCM_NONBLOCK. In the NONBLOCK case, this could be simplified to written = snd_pcm_write(frames); since I discovered that ALSA can write a little more than what avail returns, and NONBLOCK implies it'll not wait in an attempt to write a 2s data buffer, but return written < frames instead. Actually, I've a slightly more elaborate sequence in mind. I've read that snd_pcm_open() may be delayed because of networking issues, which I want to avoid in the audio thread. Therefore I'm considering using: snd_pcm_open(SND_PCM_NONBLOCK); ... setup hw&sw_params snd_pcm_prepare() snd_pcm_nonblock(0); /* or should it be called before prepare? */ I.e. allow blocking while playing, which does not actually happen thanks to avail_update() and a push model based on periodic timers to feed data with pcm_write(). Note that so far, I've not questioned whether a timer-based push model actually makes sense for Wine with ALSA... Topic T4 mmap After I re-read the mmdevapi documentation, I believe that mmdevapi's GetBuffer/ReleaseBuffer rendering protocol may be compatible with ALSA's snd_pcm_mmap_begin+commit after all. It all depends on whether ALSA grants requests for sizes as large as buffer_size, with buffer_size up to 2 seconds * samples/sec. Hence, as an optimization, one could imagine a driver using mmap, like Winealsa used to have prior to the 2011 rewrite. Yet it always had the fallback without mmap, so let's concentrate on the non-mmap case initially and get that right. Thank you for your help and for reading up to this point, Jörg Höhle

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Re: [alsa-devel] USB Audio questions
by David Henningsson 16 Aug '11

16 Aug '11

On 08/15/2011 10:31 PM, Pierre-Louis Bossart wrote: > - Increasing the number of packets/urbs solves my power issue but not the > synchronization issue. If I reduce the number of urbs to reduce the > interrupt rate, then the accuracy of the hw_pointer is decreased big time > and it becomes difficult to synchronize with video. I think the same thing is a problem for quite a few other devices as well - I wonder if we need some kind of "pointer granularity" variable to be exported through the ALSA API? PulseAudio could use that to determine whether or not to enable timer-based scheduling. And in these cases, maybe a call to hw_pointer could return hw_pointer and time, and then PulseAudio etc could use that for extrapolation (or the extrapolation could be done in alsa-lib). -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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[alsa-devel] [PATCH] ASoC: Implement WM8994 thermal warning and shutdown interrupt support
by Mark Brown 16 Aug '11

16 Aug '11

ALSA doesn't really have good mechanisms for dealing with these so we just log them - the hardware already has automatic shutdown support. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 26 ++++++++++++++++++++++++++ 1 files changed, 26 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index a0d6274..9412491 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3045,6 +3045,24 @@ static irqreturn_t wm8994_fifo_error(int irq, void *data) return IRQ_HANDLED; } +static irqreturn_t wm8994_temp_warn(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + + dev_err(codec->dev, "Thermal warning\n"); + + return IRQ_HANDLED; +} + +static irqreturn_t wm8994_temp_shut(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + + dev_crit(codec->dev, "Thermal shutdown\n"); + + return IRQ_HANDLED; +} + static int wm8994_codec_probe(struct snd_soc_codec *codec) { struct wm8994 *control; @@ -3123,6 +3141,10 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994_request_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, wm8994_fifo_error, "FIFO error", codec); + wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_TEMP_WARN, + wm8994_temp_warn, "Thermal warning", codec); + wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_TEMP_SHUT, + wm8994_temp_shut, "Thermal shutdown", codec); ret = wm8994_request_irq(codec->control_data, WM8994_IRQ_DCS_DONE, wm_hubs_dcs_done, "DC servo done", @@ -3387,6 +3409,8 @@ err_irq: wm8994_free_irq(codec->control_data, WM8994_IRQ_DCS_DONE, &wm8994->hubs); wm8994_free_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, codec); + wm8994_free_irq(codec->control_data, WM8994_IRQ_TEMP_SHUT, codec); + wm8994_free_irq(codec->control_data, WM8994_IRQ_TEMP_WARN, codec); err: kfree(wm8994); return ret; @@ -3409,6 +3433,8 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) wm8994_free_irq(codec->control_data, WM8994_IRQ_DCS_DONE, &wm8994->hubs); wm8994_free_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, codec); + wm8994_free_irq(codec->control_data, WM8994_IRQ_TEMP_SHUT, codec); + wm8994_free_irq(codec->control_data, WM8994_IRQ_TEMP_WARN, codec); switch (control->type) { case WM8994: -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: Add WM8958 noise gate support
by Mark Brown 16 Aug '11

16 Aug '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/linux/mfd/wm8994/registers.h | 45 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8994-tables.c | 12 ++++---- sound/soc/codecs/wm8994.c | 38 ++++++++++++++++++++++++++++ 3 files changed, 89 insertions(+), 6 deletions(-) diff --git a/include/linux/mfd/wm8994/registers.h b/include/linux/mfd/wm8994/registers.h index 54dbaa9..d7be463 100644 --- a/include/linux/mfd/wm8994/registers.h +++ b/include/linux/mfd/wm8994/registers.h @@ -138,6 +138,8 @@ #define WM8994_AIF1_DAC1_FILTERS_2 0x421 #define WM8994_AIF1_DAC2_FILTERS_1 0x422 #define WM8994_AIF1_DAC2_FILTERS_2 0x423 +#define WM8958_AIF1_DAC1_NOISE_GATE 0x430 +#define WM8958_AIF1_DAC2_NOISE_GATE 0x431 #define WM8994_AIF1_DRC1_1 0x440 #define WM8994_AIF1_DRC1_2 0x441 #define WM8994_AIF1_DRC1_3 0x442 @@ -195,6 +197,7 @@ #define WM8994_AIF2_ADC_FILTERS 0x510 #define WM8994_AIF2_DAC_FILTERS_1 0x520 #define WM8994_AIF2_DAC_FILTERS_2 0x521 +#define WM8958_AIF2_DAC_NOISE_GATE 0x530 #define WM8994_AIF2_DRC_1 0x540 #define WM8994_AIF2_DRC_2 0x541 #define WM8994_AIF2_DRC_3 0x542 @@ -3040,6 +3043,34 @@ #define WM8994_AIF1DAC2_3D_ENA_WIDTH 1 /* AIF1DAC2_3D_ENA */ /* + * R1072 (0x430) - AIF1 DAC1 Noise Gate + */ +#define WM8958_AIF1DAC1_NG_HLD_MASK 0x0060 /* AIF1DAC1_NG_HLD - [6:5] */ +#define WM8958_AIF1DAC1_NG_HLD_SHIFT 5 /* AIF1DAC1_NG_HLD - [6:5] */ +#define WM8958_AIF1DAC1_NG_HLD_WIDTH 2 /* AIF1DAC1_NG_HLD - [6:5] */ +#define WM8958_AIF1DAC1_NG_THR_MASK 0x000E /* AIF1DAC1_NG_THR - [3:1] */ +#define WM8958_AIF1DAC1_NG_THR_SHIFT 1 /* AIF1DAC1_NG_THR - [3:1] */ +#define WM8958_AIF1DAC1_NG_THR_WIDTH 3 /* AIF1DAC1_NG_THR - [3:1] */ +#define WM8958_AIF1DAC1_NG_ENA 0x0001 /* AIF1DAC1_NG_ENA */ +#define WM8958_AIF1DAC1_NG_ENA_MASK 0x0001 /* AIF1DAC1_NG_ENA */ +#define WM8958_AIF1DAC1_NG_ENA_SHIFT 0 /* AIF1DAC1_NG_ENA */ +#define WM8958_AIF1DAC1_NG_ENA_WIDTH 1 /* AIF1DAC1_NG_ENA */ + +/* + * R1073 (0x431) - AIF1 DAC2 Noise Gate + */ +#define WM8958_AIF1DAC2_NG_HLD_MASK 0x0060 /* AIF1DAC2_NG_HLD - [6:5] */ +#define WM8958_AIF1DAC2_NG_HLD_SHIFT 5 /* AIF1DAC2_NG_HLD - [6:5] */ +#define WM8958_AIF1DAC2_NG_HLD_WIDTH 2 /* AIF1DAC2_NG_HLD - [6:5] */ +#define WM8958_AIF1DAC2_NG_THR_MASK 0x000E /* AIF1DAC2_NG_THR - [3:1] */ +#define WM8958_AIF1DAC2_NG_THR_SHIFT 1 /* AIF1DAC2_NG_THR - [3:1] */ +#define WM8958_AIF1DAC2_NG_THR_WIDTH 3 /* AIF1DAC2_NG_THR - [3:1] */ +#define WM8958_AIF1DAC2_NG_ENA 0x0001 /* AIF1DAC2_NG_ENA */ +#define WM8958_AIF1DAC2_NG_ENA_MASK 0x0001 /* AIF1DAC2_NG_ENA */ +#define WM8958_AIF1DAC2_NG_ENA_SHIFT 0 /* AIF1DAC2_NG_ENA */ +#define WM8958_AIF1DAC2_NG_ENA_WIDTH 1 /* AIF1DAC2_NG_ENA */ + +/* * R1088 (0x440) - AIF1 DRC1 (1) */ #define WM8994_AIF1DRC1_SIG_DET_RMS_MASK 0xF800 /* AIF1DRC1_SIG_DET_RMS - [15:11] */ @@ -3651,6 +3682,20 @@ #define WM8994_AIF2DAC_3D_ENA_WIDTH 1 /* AIF2DAC_3D_ENA */ /* + * R1328 (0x530) - AIF2 DAC Noise Gate + */ +#define WM8958_AIF2DAC_NG_HLD_MASK 0x0060 /* AIF2DAC_NG_HLD - [6:5] */ +#define WM8958_AIF2DAC_NG_HLD_SHIFT 5 /* AIF2DAC_NG_HLD - [6:5] */ +#define WM8958_AIF2DAC_NG_HLD_WIDTH 2 /* AIF2DAC_NG_HLD - [6:5] */ +#define WM8958_AIF2DAC_NG_THR_MASK 0x000E /* AIF2DAC_NG_THR - [3:1] */ +#define WM8958_AIF2DAC_NG_THR_SHIFT 1 /* AIF2DAC_NG_THR - [3:1] */ +#define WM8958_AIF2DAC_NG_THR_WIDTH 3 /* AIF2DAC_NG_THR - [3:1] */ +#define WM8958_AIF2DAC_NG_ENA 0x0001 /* AIF2DAC_NG_ENA */ +#define WM8958_AIF2DAC_NG_ENA_MASK 0x0001 /* AIF2DAC_NG_ENA */ +#define WM8958_AIF2DAC_NG_ENA_SHIFT 0 /* AIF2DAC_NG_ENA */ +#define WM8958_AIF2DAC_NG_ENA_WIDTH 1 /* AIF2DAC_NG_ENA */ + +/* * R1344 (0x540) - AIF2 DRC (1) */ #define WM8994_AIF2DRC_SIG_DET_RMS_MASK 0xF800 /* AIF2DRC_SIG_DET_RMS - [15:11] */ diff --git a/sound/soc/codecs/wm8994-tables.c b/sound/soc/codecs/wm8994-tables.c index 428d4c5..43e6603 100644 --- a/sound/soc/codecs/wm8994-tables.c +++ b/sound/soc/codecs/wm8994-tables.c @@ -1073,8 +1073,8 @@ const struct wm8994_access_mask wm8994_access_masks[WM8994_CACHE_SIZE] = { { 0x0000, 0x0000 }, /* R1069 */ { 0x0000, 0x0000 }, /* R1070 */ { 0x0000, 0x0000 }, /* R1071 */ - { 0x0000, 0x0000 }, /* R1072 */ - { 0x0000, 0x0000 }, /* R1073 */ + { 0x006F, 0x006F }, /* R1072 - AIF1 DAC1 Noise Gate */ + { 0x006F, 0x006F }, /* R1073 - AIF1 DAC2 Noise Gate */ { 0x0000, 0x0000 }, /* R1074 */ { 0x0000, 0x0000 }, /* R1075 */ { 0x0000, 0x0000 }, /* R1076 */ @@ -1329,7 +1329,7 @@ const struct wm8994_access_mask wm8994_access_masks[WM8994_CACHE_SIZE] = { { 0x0000, 0x0000 }, /* R1325 */ { 0x0000, 0x0000 }, /* R1326 */ { 0x0000, 0x0000 }, /* R1327 */ - { 0x0000, 0x0000 }, /* R1328 */ + { 0x006F, 0x006F }, /* R1328 - AIF2 DAC Noise Gate */ { 0x0000, 0x0000 }, /* R1329 */ { 0x0000, 0x0000 }, /* R1330 */ { 0x0000, 0x0000 }, /* R1331 */ @@ -2646,8 +2646,8 @@ const u16 wm8994_reg_defaults[WM8994_CACHE_SIZE] = { 0x0000, /* R1069 */ 0x0000, /* R1070 */ 0x0000, /* R1071 */ - 0x0000, /* R1072 */ - 0x0000, /* R1073 */ + 0x0068, /* R1072 - AIF1 DAC1 Noise Gate */ + 0x0068, /* R1073 - AIF1 DAC2 Noise Gate */ 0x0000, /* R1074 */ 0x0000, /* R1075 */ 0x0000, /* R1076 */ @@ -2902,7 +2902,7 @@ const u16 wm8994_reg_defaults[WM8994_CACHE_SIZE] = { 0x0000, /* R1325 */ 0x0000, /* R1326 */ 0x0000, /* R1327 */ - 0x0000, /* R1328 */ + 0x0068, /* R1328 - AIF2 DAC Noise Gate */ 0x0000, /* R1329 */ 0x0000, /* R1330 */ 0x0000, /* R1331 */ diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index f328369..c45b634 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -299,6 +299,7 @@ static const DECLARE_TLV_DB_SCALE(st_tlv, -3600, 300, 0); static const DECLARE_TLV_DB_SCALE(wm8994_3d_tlv, -1600, 183, 0); static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); static const DECLARE_TLV_DB_SCALE(mixin_boost_tlv, 0, 900, 0); +static const DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0); #define WM8994_DRC_SWITCH(xname, reg, shift) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ @@ -678,8 +679,45 @@ SOC_SINGLE_TLV("AIF2 EQ5 Volume", WM8994_AIF2_EQ_GAINS_2, 6, 31, 0, eq_tlv), }; +static const char *wm8958_ng_text[] = { + "30ms", "125ms", "250ms", "500ms", +}; + +static const struct soc_enum wm8958_aif1dac1_ng_hold = + SOC_ENUM_SINGLE(WM8958_AIF1_DAC1_NOISE_GATE, + WM8958_AIF1DAC1_NG_THR_SHIFT, 4, wm8958_ng_text); + +static const struct soc_enum wm8958_aif1dac2_ng_hold = + SOC_ENUM_SINGLE(WM8958_AIF1_DAC2_NOISE_GATE, + WM8958_AIF1DAC2_NG_THR_SHIFT, 4, wm8958_ng_text); + +static const struct soc_enum wm8958_aif2dac_ng_hold = + SOC_ENUM_SINGLE(WM8958_AIF2_DAC_NOISE_GATE, + WM8958_AIF2DAC_NG_THR_SHIFT, 4, wm8958_ng_text); + static const struct snd_kcontrol_new wm8958_snd_controls[] = { SOC_SINGLE_TLV("AIF3 Boost Volume", WM8958_AIF3_CONTROL_2, 10, 3, 0, aif_tlv), + +SOC_SINGLE("AIF1DAC1 Noise Gate Switch", WM8958_AIF1_DAC1_NOISE_GATE, + WM8958_AIF1DAC1_NG_ENA_SHIFT, 1, 0), +SOC_ENUM("AIF1DAC1 Noise Gate Hold Time", wm8958_aif1dac1_ng_hold), +SOC_SINGLE_TLV("AIF1DAC1 Noise Gate Threshold Volume", + WM8958_AIF1_DAC1_NOISE_GATE, WM8958_AIF1DAC1_NG_THR_SHIFT, + 7, 1, ng_tlv), + +SOC_SINGLE("AIF1DAC2 Noise Gate Switch", WM8958_AIF1_DAC2_NOISE_GATE, + WM8958_AIF1DAC2_NG_ENA_SHIFT, 1, 0), +SOC_ENUM("AIF1DAC2 Noise Gate Hold Time", wm8958_aif1dac2_ng_hold), +SOC_SINGLE_TLV("AIF1DAC2 Noise Gate Threshold Volume", + WM8958_AIF1_DAC2_NOISE_GATE, WM8958_AIF1DAC2_NG_THR_SHIFT, + 7, 1, ng_tlv), + +SOC_SINGLE("AIF2DAC Noise Gate Switch", WM8958_AIF2_DAC_NOISE_GATE, + WM8958_AIF2DAC_NG_ENA_SHIFT, 1, 0), +SOC_ENUM("AIF2DAC Noise Gate Hold Time", wm8958_aif2dac_ng_hold), +SOC_SINGLE_TLV("AIF2DAC Noise Gate Threshold Volume", + WM8958_AIF2_DAC_NOISE_GATE, WM8958_AIF2DAC_NG_THR_SHIFT, + 7, 1, ng_tlv), }; static const struct snd_kcontrol_new wm1811_snd_controls[] = { -- 1.7.5.4

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Re: [alsa-devel] USB Audio questions
by Daniel Mack 16 Aug '11

16 Aug '11

Hi, On Mon, Aug 15, 2011 at 10:31 PM, Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> wrote: > I am trying to prototype an application where the key requirement is > playback with minimal power on USB audio accessories while maintaining tight > synchronization with video. With the default ALSA USB code, I am getting > ~130 interrupts/s due to audio, a fairly high CPU load and a very poor sync, > exactly the opposite of my goals... > After a careful analysis of the code I found a couple of issues where the > help of experts would be welcome: > > - Is there any good reason why the max number of packets per urb defaults to > 8? This end-ups being the root cause for the large number of interrupts. My > headset supports up to 192 bytes/packet, that means 1536 bytes/urb and 125 > interrupts/s. Using more packets/urb results in a linear decrease of the > interrupt rate. You cannot use more than 8 subframes per ISO urb, that's what how USB is designed. What exactly did you do in you test? Did you increase MAX_URBS? > - Increasing the number of packets/urbs solves my power issue but not the > synchronization issue. If I reduce the number of urbs to reduce the > interrupt rate, then the accuracy of the hw_pointer is decreased big time > and it becomes difficult to synchronize with video. The .pointer routine > only keeps track of retired urbs, it seems there's no way to know how many > samples were played at a given time? It also makes pulseaudio behave > somewhat strangely since the driver only provides an approximate position. > In the CAIQ driver code, the .pointer routine seems to rely on a direct > register read, a more efficient way to do things. Isn't there a direct way > to query the usb core to know how many bytes have been transmitted and make > the hw_ptr/delay information independent of the urb buffer size? Just to be sure - you're talking about the driver in sound/usb/caiaq, right? After a short comparison, I'd say both the caiaq driver and the generic driver do more or less the same in regard to the .pointer value, as for the output stream, they both increase their internal counter when the packet is prepared, shortly before it is submitted to the host controller. Increasing the number of URBs. Admittedly, this approach is questionable, as a larger number of prepared output buffers will actually cause the client believe that more data has already been written already. You could try moving the hwptr_done increase statement into the retire callback (which is the first place we get to know about packets that have been sent to the device)? > - Is there any reason why the driver relies on vmalloc and memcpy? It seems > like using physically-contiguous memory would be more efficient, as done in > the CAIQ driver? I hacked something to use prealloc_pages/malloc_pages and > things seem to work fine. Can you show a patch? Daniel

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Re: [alsa-devel] hda-intel and runtime power-management
by Tatulea, Dragos 16 Aug '11

16 Aug '11

On Mon, Aug 8, 2011 at 9:11 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > At Sun, 7 Aug 2011 20:55:52 +0300, > Tatulea, Dragos wrote: > > > >> >> I'm in the process of enabling pm runtime pm for the hda_intel > driver, > > >> >> not being sure if this is a sane task :). Noticed that > > >> >> CONFIG_SND_HDA_POWER_SAVE disables codecs and chip when they are > not > > >> >> in use, but the power gain is not noticeable on my hw. On the other > > >> >> hand, a full suspend (azx_suspend) does save a few hundred mW's. > > >> >> I am aware that this is hw specific, but I would like to take > > >> >> advantage of the full suspend. Suspend/resume latency is below > jiffy > > >> >> resolution. > > >> > > > >> > Hm, it's strange. Basically the current power-save does almost > > >> > equivalent with the normal suspend/resume, i.e. the codec is > > >> > suspended and azx_stop_chip() is called. Make sure that the > > >> > power-saving is really kicked in during your test. > > >> > > > >> Indeed, I checked and a codec is stuck in a power transition for some > reason. > > > > > > You should check who accessing the device files via fuser, etc. > > > > Yes, there's a non-muted pcm around there that prevents the codec from sleeping. I have a patch that checks for all substream states and puts the chip to sleep when they are all paused/suspended. Is this a bad idea? It could be that resuming the chip takes too long, depending on hw... > > >> Besides that, it would be nicer to have the currently standard runtime > > >> pm on codec bus instead of doing the same thing with > > >> CONFIG_SND_HDA_POWER_SAVE, right? > > > > > > Ideally, yes, but practically it's difficult, so far, because the > > > driver can manage multiple codecs and we need the individual codec > > > power-up/down. With the current runtime pm scheme, it's difficult to > > > implement like this, as it's no simple suspend/resume of the driver > > > itself. > > > > > The codec bus can contain the generic runtime pm part which will call > > runtime_suspend/_resume for individual codecs, depending on usage. > > Each codec can register it's own dev_pm_ops. > > Yes, but this would need relatively big changes in the structure > of the hd-audio driver, e.g. assigning struct device per codec > instance, etc. > > Still looking into it, maybe there's a less invasive way of doing this. Thanks, Dragos

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Re: [alsa-devel] HDAudio period/buffer sizes
by Takashi Iwai 15 Aug '11

15 Aug '11

At Tue, 26 Jul 2011 14:23:40 -0500, Pierre-Louis Bossart wrote: > > that the buffer itself was already aligned to 128 bytes from the > > very beginning as it's allocated via dma_alloc_coherent(). > > The fact that the commit fixes the real problem means that the buffer > > size must be aligned, too, at least for some hardware. > > I checked internally and the reply was "There is no requirement the buffer > must be 128 byte multiples. As long as it is integer number of samples, we > are good." > > What's the best way to deal with it? Add a driver option? A compilation > option? I'd say a driver option since this must be dynamically adjustable. Care to make a patch? For 3.2 (after 3.1 merge window), we can disable 128-bytes limit as default again for testing. If someone hits the problem, he can switch easily and we can investigate more at this time. thanks, Takashi

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Re: [alsa-devel] [PATCH] ASoC: sta32x: shortcut the for loop to get ir and mcs
by Mark Brown 15 Aug '11

15 Aug '11

On Sun, Aug 14, 2011 at 11:31:04AM +0800, Axel Lin wrote: > There is exactly one match or no match at all during the for loop iteration, > thus we can break from the for loop once a match is found. > > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> Applied, thanks.

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Re: [alsa-devel] [PATCH] ASoC: nuc900-pcm: remove unused variable 'dai'
by Mark Brown 15 Aug '11

15 Aug '11

On Mon, Aug 15, 2011 at 04:38:34PM +0800, Axel Lin wrote: > Remove unused variable 'dai' to eliminate below warning. Applied, thanks.

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Re: [alsa-devel] [PATCH] ASoC: soc-pcm: Remove unused global mutex
by Mark Brown 15 Aug '11

15 Aug '11

On Sun, Aug 14, 2011 at 08:00:12AM +0800, Axel Lin wrote: > Since commit b8c0dab9bf3373010e857a8d3f1b594c60a348dd > "ASoC: core - PCM mutex per rtd", > the global pcm_mutex is not being used any more. Applied, thanks.

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Re: [alsa-devel] [PATCH] ASoC: soc-cache: Remove unneeded codec_drv pointer variable in snd_soc_lzo_get_blksize
by Mark Brown 15 Aug '11

15 Aug '11

On Sat, Aug 13, 2011 at 09:27:16PM +0800, Axel Lin wrote: > Since commit aea170a099793abcd0e6de46b947458073204241 > "ASoC: soc-cache: Add reg_size as a member to snd_soc_codec", > the codec_drv pointer variable is not used in snd_soc_lzo_get_blksize. Applied, thanks.

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Re: [alsa-devel] [PATCH] ASoC: soc-cache: Remove unneeded codec_drv pointer variable in snd_soc_flat_cache_init
by Mark Brown 15 Aug '11

15 Aug '11

On Sat, Aug 13, 2011 at 09:12:01PM +0800, Axel Lin wrote: > Since commit d779fce5d79525d66269c8f6e430e1515d697f3d > "ASoC: soc-cache: Ensure flat compression uses a copy of the defaults cache", > the codec_drv pointer variable is not used any more. Applied, thanks.

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Re: [alsa-devel] [PATCH] ASoC: s6000-pcm: remove unused variable 'dai'
by Mark Brown 15 Aug '11

15 Aug '11

On Sat, Aug 13, 2011 at 11:32:56AM +0800, Axel Lin wrote: > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> Applied, thanks.

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[alsa-devel] [PATCH v2 1/5] ASoC: ad193x: fix registers definition
by Scott Jiang 15 Aug '11

15 Aug '11

fix dac word len mask and adc tdm fmt shift value Signed-off-by: Scott Jiang <scott.jiang.linux(a)gmail.com> --- sound/soc/codecs/ad193x.h | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h index 9747b54..c1029d2 100644 --- a/sound/soc/codecs/ad193x.h +++ b/sound/soc/codecs/ad193x.h @@ -34,7 +34,7 @@ #define AD193X_DAC_LEFT_HIGH (1 << 3) #define AD193X_DAC_BCLK_INV (1 << 7) #define AD193X_DAC_CTRL2 0x804 -#define AD193X_DAC_WORD_LEN_MASK 0xC +#define AD193X_DAC_WORD_LEN_MASK 0x18 #define AD193X_DAC_MASTER_MUTE 1 #define AD193X_DAC_CHNL_MUTE 0x805 #define AD193X_DACL1_MUTE 0 @@ -63,7 +63,7 @@ #define AD193X_ADC_CTRL1 0x80f #define AD193X_ADC_SERFMT_MASK 0x60 #define AD193X_ADC_SERFMT_STEREO (0 << 5) -#define AD193X_ADC_SERFMT_TDM (1 << 2) +#define AD193X_ADC_SERFMT_TDM (1 << 5) #define AD193X_ADC_SERFMT_AUX (2 << 5) #define AD193X_ADC_WORD_LEN_MASK 0x3 #define AD193X_ADC_CTRL2 0x810 -- 1.7.0.4

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