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

[alsa-devel] [PATCH 1/4] sound/soc/kirkwood/kirkwood-i2s.c: add missing kfree
by Julia Lawall 12 Sep '11

12 Sep '11

From: Julia Lawall <julia(a)diku.dk> Adjust the goto to jump to the error handling code that includes kfree. 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/kirkwood/kirkwood-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index a33fc51..8f16cd3 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -424,7 +424,7 @@ static __devinit int kirkwood_i2s_dev_probe(struct platform_device *pdev) if (!priv->mem) { dev_err(&pdev->dev, "request_mem_region failed\n"); err = -EBUSY; - goto error; + goto error_alloc; } priv->io = ioremap(priv->mem->start, SZ_16K);

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[alsa-devel] Problem with sound on Asus EeePC 1215b
by Touko Korpela 12 Sep '11

12 Sep '11

(I had to subscribe to this list, otherwise mails seem to stuck forever, moderation doesn't seem to work. But please reply to me directly, I don't receive messages sent to the list, because I disabled that in settings) Your bug tracker is useless now, because driver maintainers don't use it. It's not nice to report bugs there if nobody reads them. www.alsa-project.org /db/?f=e13caeff15a01968117053a7c7fe9588715aec8b Support for this hardware seem to have some issues. No sound is heard (but mixer is not muted). Shouldn't mixer have more channels to adjust? Linux kernel version is 3.0.0-1-amd64 (Debian 3.0.0-3), containing upstream 3.0.3

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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

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[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

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[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

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[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

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[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(-)

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[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

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[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

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