February 2012 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] Random process lockup on ARM board: alsa-lib-1.0.25, FUTEX_WAIT_PRIVATE
by Clemens Ladisch 20 Feb '12

20 Feb '12

Jonathan Andrews wrote: > I've tried building alsalib in many ways, at the moment its : > $ ./configure --target=arm-linux --host=i686-linux --enable-shared --disable-static --without-pthread Try adding --disable-ucm and --disable-python. Regards, Clemens

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[alsa-devel] ASoC: fsi: Add FSI DMAEngine support
by Kuninori Morimoto 20 Feb '12

20 Feb '12

Hi Mark, Liam, and Paul These are DMAEngine support for FSI driver I created a lot of patches since FSI DMAEngine needed much modification. I hope it was good for reviewing. But some patches are just cleanup/tidyup/rename. Last patch is main DMAEngine patch. it support Tx case only at this point. FSI will use PIO version if CPU/platform doesn't have DMAEngine support. >> Paul Could you please check #16 patch which modify arch/arm/mach-shmobile and arch/sh Kuninori Morimoto (19): ASoC: fsi: reduce runtime calculation by using pre-setting ASoC: fsi: tidyup: fsi_stream_xx() functions were gathered ASoC: fsi: data push/pop calculation part was divided ASoC: fsi: rename fsi_dma_soft_xxx() to fsi_pio_xxx() ASoC: fsi: tidyup: move fsi_fifo_init() onto fsi_hw_startup() ASoC: fsi: remove unnecessary parameter from fsi_hw_shutdown() ASoC: fsi: rename fsi_stream_push/pop() to fsi_stream_init/quit() ASoC: fsi: modify fsi_pio_get_area() parameter and using position ASoC: fsi: re-define fsi_is_play() and fsi_stream_is_play() ASoC: fsi: use fsi_stream in fsi_get_current_fifo_samples() parameter ASoC: fsi: add fsi_stream_handler and PIO handler ASoC: fsi: tidyup: fsi_pio_xxx() are gathered ASoC: fsi: don't use is_play as a parameter of fsi functions ASoC: fsi: add .start_stop handler to fsi_stream_handler ASoC: fsi: fsi_stream_is_working() care substream->runtime ASoC: fsi: PortA/B information was controlled by sh_fsi_port_info ASoC: fsi: add .init/.quit handler support ASoC: fsi: fixup fsi_pointer() calculation method ASoC: fsi: Add DMAEngine support Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH] ASoC: Add __devinit annotation for pxa2xx_ac97_probe
by Axel Lin 20 Feb '12

20 Feb '12

This fixes below build warning: WARNING: vmlinux.o(.text+0x1e632c): Section mismatch in reference from the function pxa2xx_ac97_probe() to the function .devinit.text:pxa2xx_ac97_hw_probe() The function pxa2xx_ac97_probe() references the function __devinit pxa2xx_ac97_hw_probe(). This is often because pxa2xx_ac97_probe lacks a __devinit annotation or the annotation of pxa2xx_ac97_hw_probe is wrong. Also rename pxa_ac97_dai to pxa_ac97_dai_driver to fix below build warning: LD sound/soc/pxa/built-in.o WARNING: sound/soc/pxa/built-in.o(.data+0x18c): Section mismatch in reference from the variable pxa_ac97_dai to the function .devinit.text:pxa2xx_ac97_probe() The variable pxa_ac97_dai references the function __devinit pxa2xx_ac97_probe() If the reference is valid then annotate the variable with __init* or __refdata (see linux/init.h) or name the variable: *driver, *_template, *_timer, *_sht, *_ops, *_probe, *_probe_one, *_console Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/pxa/pxa2xx-ac97.c | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 837ff34..4800d5f 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -103,7 +103,7 @@ static int pxa2xx_ac97_resume(struct snd_soc_dai *dai) #define pxa2xx_ac97_resume NULL #endif -static int pxa2xx_ac97_probe(struct snd_soc_dai *dai) +static int __devinit pxa2xx_ac97_probe(struct snd_soc_dai *dai) { return pxa2xx_ac97_hw_probe(to_platform_device(dai->dev)); } @@ -179,7 +179,7 @@ static const struct snd_soc_dai_ops pxa_ac97_mic_dai_ops = { * There is only 1 physical AC97 interface for pxa2xx, but it * has extra fifo's that can be used for aux DACs and ADCs. */ -static struct snd_soc_dai_driver pxa_ac97_dai[] = { +static struct snd_soc_dai_driver pxa_ac97_dai_driver[] = { { .name = "pxa2xx-ac97", .ac97_control = 1, @@ -244,13 +244,13 @@ static __devinit int pxa2xx_ac97_dev_probe(struct platform_device *pdev) * driver to do interesting things with the clocking to get us up * and running. */ - return snd_soc_register_dais(&pdev->dev, pxa_ac97_dai, - ARRAY_SIZE(pxa_ac97_dai)); + return snd_soc_register_dais(&pdev->dev, pxa_ac97_dai_driver, + ARRAY_SIZE(pxa_ac97_dai_driver)); } static int __devexit pxa2xx_ac97_dev_remove(struct platform_device *pdev) { - snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(pxa_ac97_dai)); + snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(pxa_ac97_dai_driver)); return 0; } -- 1.7.5.4

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[alsa-devel] [PATCH 1/2] ALSA: Remove __devinit annotation for pxa2xx_ac97_hw_probe to fix section mismatch warning
by Axel Lin 20 Feb '12

20 Feb '12

This fixes below build warning: WARNING: vmlinux.o(.text+0x1e632c): Section mismatch in reference from the function pxa2xx_ac97_probe() to the function .devinit.text:pxa2xx_ac97_hw_probe() The function pxa2xx_ac97_probe() references the function __devinit pxa2xx_ac97_hw_probe(). This is often because pxa2xx_ac97_probe lacks a __devinit annotation or the annotation of pxa2xx_ac97_hw_probe is wrong. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/arm/pxa2xx-ac97-lib.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 0e9e822..5e926c4 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -313,7 +313,7 @@ int pxa2xx_ac97_hw_resume(void) EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_resume); #endif -int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev) +int pxa2xx_ac97_hw_probe(struct platform_device *dev) { int ret; pxa2xx_audio_ops_t *pdata = dev->dev.platform_data; -- 1.7.5.4

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[alsa-devel] [PATCH] checkpatch fixes in ASoC
by Vikram Narayanan 19 Feb '12

19 Feb '12

Various checkpatch fixes in the ASoC tree Signed-off-by: Vikram Narayanan <vikram186(a)gmail.com> --- sound/soc/soc-cache.c | 5 ++--- sound/soc/soc-core.c | 17 ++++++++--------- sound/soc/soc-dapm.c | 14 ++++++++------ sound/soc/soc-pcm.c | 4 ++-- sound/soc/soc-utils.c | 2 +- 5 files changed, 21 insertions(+), 21 deletions(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 9077aa4..6b6c226 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -73,7 +73,7 @@ struct snd_soc_rbtree_node { unsigned int word_size; /* number of bytes needed to represent the register index */ void *block; /* block of adjacent registers */ unsigned int blklen; /* number of registers available in the block */ -} __attribute__ ((packed)); +} __attribute__ ((__packed)); struct snd_soc_rbtree_ctx { struct rb_root root; @@ -1017,9 +1017,8 @@ int snd_soc_cache_sync(struct snd_soc_codec *codec) int ret; const char *name; - if (!codec->cache_sync) { + if (!codec->cache_sync) return 0; - } if (!codec->cache_ops || !codec->cache_ops->sync) return -ENOSYS; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a25fa63..c13d414 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -195,7 +195,7 @@ static ssize_t pmdown_time_set(struct device *dev, container_of(dev, struct snd_soc_pcm_runtime, dev); int ret; - ret = strict_strtol(buf, 10, &rtd->pmdown_time); + ret = kstrtol(buf, 10, &rtd->pmdown_time); if (ret) return ret; @@ -254,10 +254,10 @@ static ssize_t codec_reg_write_file(struct file *file, while (*start == ' ') start++; - reg = simple_strtoul(start, &start, 16); + reg = kstrtoul(start, &start, 16); while (*start == ' ') start++; - if (strict_strtoul(start, 16, &value)) + if (kstrtoul(start, 16, &value)) return -EINVAL; /* Userspace has been fiddling around behind the kernel's back */ @@ -458,7 +458,7 @@ static int soc_ac97_dev_unregister(struct snd_soc_codec *codec) } /* stop no dev release warning */ -static void soc_ac97_device_release(struct device *dev){} +static void soc_ac97_device_release(struct device *dev) {} /* register ac97 codec to bus */ static int soc_ac97_dev_register(struct snd_soc_codec *codec) @@ -758,9 +758,9 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) dev_dbg(card->dev, "binding %s at idx %d\n", dai_link->name, num); /* do we already have the CPU DAI for this link ? */ - if (rtd->cpu_dai) { + if (rtd->cpu_dai) goto find_codec; - } + /* no, then find CPU DAI from registered DAIs*/ list_for_each_entry(cpu_dai, &dai_list, list) { if (!strcmp(cpu_dai->name, dai_link->cpu_dai_name)) { @@ -773,9 +773,8 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) find_codec: /* do we already have the CODEC for this link ? */ - if (rtd->codec) { + if (rtd->codec) goto find_platform; - } /* no, then find CODEC from registered CODECs*/ list_for_each_entry(codec, &codec_list, list) { @@ -2028,7 +2027,7 @@ int snd_soc_add_platform_controls(struct snd_soc_platform *platform, err = snd_ctl_add(card, snd_soc_cnew(control, platform, control->name, NULL)); if (err < 0) { - dev_err(platform->dev, "Failed to add %s %d\n",control->name, err); + dev_err(platform->dev, "Failed to add %s %d\n", control->name, err); return err; } } diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index fcd4cb5..07abd58 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -379,7 +379,7 @@ static int dapm_connect_mux(struct snd_soc_dapm_context *dapm, list_add(&path->list, &dapm->card->paths); list_add(&path->list_sink, &dest->sources); list_add(&path->list_source, &src->sinks); - path->name = (char*)e->texts[i]; + path->name = (char *)e->texts[i]; dapm_set_path_status(dest, path, 0); return 0; } @@ -1769,7 +1769,7 @@ static ssize_t dapm_widget_show(struct device *dev, { struct snd_soc_pcm_runtime *rtd = container_of(dev, struct snd_soc_pcm_runtime, dev); - struct snd_soc_codec *codec =rtd->codec; + struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_widget *w; int count = 0; char *state = "not set"; @@ -1795,7 +1795,7 @@ static ssize_t dapm_widget_show(struct device *dev, case snd_soc_dapm_regulator_supply: if (w->name) count += sprintf(buf + count, "%s: %s\n", - w->name, w->power ? "On":"Off"); + w->name, w->power ? "On" : "Off"); break; default: break; @@ -2209,7 +2209,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) return -ENOMEM; } - switch(w->id) { + switch (w->id) { case snd_soc_dapm_switch: case snd_soc_dapm_mixer: case snd_soc_dapm_mixer_named_ctl: @@ -2700,7 +2700,9 @@ int snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, size_t name_len; int ret; - if ((w = dapm_cnew_widget(widget)) == NULL) + w = dapm_cnew_widget(widget); + + if (!w) return -ENOMEM; switch (w->id) { @@ -2829,7 +2831,7 @@ static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm, w->name, w->sname, stream, event); if (strstr(w->sname, stream)) { dapm_mark_dirty(w, "stream event"); - switch(event) { + switch (event) { case SND_SOC_DAPM_STREAM_START: w->active = 1; break; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index ee15337..174fdbb 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -7,7 +7,7 @@ * Copyright (C) 2010 Texas Instruments Inc. * * Authors: Liam Girdwood <lrg(a)ti.com> - * Mark Brown <broonie(a)opensource.wolfsonmicro.com> + * Mark Brown <broonie(a)opensource.wolfsonmicro.com> * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the @@ -613,7 +613,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) if (codec_dai->driver->capture.channels_min) capture = 1; - dev_dbg(rtd->card->dev, "registered pcm #%d %s\n",num,new_name); + dev_dbg(rtd->card->dev, "registered pcm #%d %s\n", num, new_name); ret = snd_pcm_new(rtd->card->snd_card, new_name, num, playback, capture, &pcm); if (ret < 0) { diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 4220bb0..2eac245 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -5,7 +5,7 @@ * * Author: Mark Brown <broonie(a)opensource.wolfsonmicro.com> * Liam Girdwood <lrg(a)slimlogic.co.uk> - * + * * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the -- 1.7.4.1

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[alsa-devel] Random process lockup on ARM board: alsa-lib-1.0.25, FUTEX_WAIT_PRIVATE
by Jonathan Andrews 18 Feb '12

18 Feb '12

Sorry all me again. I've written a program that picks up UDP broadcast packets containing audio from a number of sources, mixes it together and presents it to alsa for playback. I'm experiencing random freezes. I cant reproduce then running the process under strace as the process on the ARM board slows down to the point it underruns with ALSA all the time and the symptom doesn't show. If I attach strace after the process has frozen I get this and nothing else: # strace -p 443 Process 443 attached - interrupt to quit futex(0x4030e144, FUTEX_WAIT_PRIVATE, 2, NULL This stop happens at random at any point typically after minutes but run time can be over an hour. I need to work around this as I need the board for a demo. Im using, alsa-lib-1.0.25 kernel 2.6.36.4 BusyBox v1.19.2 I also compiled strace for arm, but this and a busybox shell is the only environment I have on the board. The problem showed when I starting using this function to poll alsa: The same code on my PC dual core AMD does not seem to suffer the problem : alsa-lib-1.0.23, Linux jonspc 2.6.32.26-175.fc12.i686 #1 SMP Before anyone points it out I want to poll as alsa is not the only sound output supported by the code. int sound_system_needs_data_now() { int framesleft; snd_pcm_status_t *status; snd_pcm_status_alloca( &status ); // If the alsa playback has stalled then it definately needs data snd_pcm_status(playback_handle,status); if( snd_pcm_status_get_state( status ) != SND_PCM_STATE_RUNNING ) return(TRUE); framesleft=snd_pcm_avail (playback_handle); //printf("FL=%d\n",framesleft); fflush(stdout); if (framesleft>1000) return(TRUE); else return(FALSE); } Stuck process # cat /proc/443/status Name: udp-many-way-au State: S (sleeping) Tgid: 443 Pid: 443 PPid: 392 TracerPid: 0 Uid: 0 0 0 0 Gid: 0 0 0 0 FDSize: 32 Groups: VmPeak: 4356 kB VmSize: 4292 kB VmLck: 132 kB VmHWM: 1788 kB VmRSS: 1788 kB VmData: 396 kB VmStk: 136 kB VmExe: 28 kB VmLib: 3348 kB VmPTE: 12 kB VmSwap: 0 kB Threads: 1 SigQ: 0/471 SigPnd: 0000000000000000 ShdPnd: 0000000000000000 SigBlk: 0000000000002000 SigIgn: 0000000000000006 SigCgt: 0000000180002000 CapInh: 0000000000000000 CapPrm: ffffffffffffffff CapEff: ffffffffffffffff CapBnd: ffffffffffffffff Cpus_allowed: 1 Cpus_allowed_list: 0 Mems_allowed: 1 Mems_allowed_list: 0 voluntary_ctxt_switches: 5908 nonvoluntary_ctxt_switches: 1766 Any useful suggestions welcome. Thanks, Jon

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[alsa-devel] [PATCH] ASoC: soc-core: Show the returned values on error messages
by Fabio Estevam 17 Feb '12

17 Feb '12

Showing the returned values on error messages is useful information. While at it, use pr_err/pr_warn whenever possible. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/soc-core.c | 39 ++++++++++++++++++++++----------------- 1 files changed, 22 insertions(+), 17 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6bad7cd..7001fb3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -931,7 +931,8 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order) if (codec_dai->driver->remove) { err = codec_dai->driver->remove(codec_dai); if (err < 0) - printk(KERN_ERR "asoc: failed to remove %s\n", codec_dai->name); + pr_err("asoc: failed to remove %s: %d\n", + codec_dai->name, err); } codec_dai->probed = 0; list_del(&codec_dai->card_list); @@ -943,7 +944,8 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order) if (platform->driver->remove) { err = platform->driver->remove(platform); if (err < 0) - printk(KERN_ERR "asoc: failed to remove %s\n", platform->name); + pr_err("asoc: failed to remove %s: %d\n", + platform->name, err); } /* Make sure all DAPM widgets are freed */ @@ -966,7 +968,8 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order) if (cpu_dai->driver->remove) { err = cpu_dai->driver->remove(cpu_dai); if (err < 0) - printk(KERN_ERR "asoc: failed to remove %s\n", cpu_dai->name); + pr_err("asoc: failed to remove %s: %d\n", + cpu_dai->name, err); } cpu_dai->probed = 0; list_del(&cpu_dai->card_list); @@ -1213,8 +1216,8 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) if (cpu_dai->driver->probe) { ret = cpu_dai->driver->probe(cpu_dai); if (ret < 0) { - printk(KERN_ERR "asoc: failed to probe CPU DAI %s\n", - cpu_dai->name); + pr_err("asoc: failed to probe CPU DAI %s: %d\n", + cpu_dai->name, ret); module_put(cpu_dai->dev->driver->owner); return ret; } @@ -1245,8 +1248,8 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) if (codec_dai->driver->probe) { ret = codec_dai->driver->probe(codec_dai); if (ret < 0) { - printk(KERN_ERR "asoc: failed to probe CODEC DAI %s\n", - codec_dai->name); + pr_err("asoc: failed to probe CODEC DAI %s: %d\n", + codec_dai->name, ret); return ret; } } @@ -1266,12 +1269,13 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) ret = device_create_file(rtd->dev, &dev_attr_pmdown_time); if (ret < 0) - printk(KERN_WARNING "asoc: failed to add pmdown_time sysfs\n"); + pr_warn("asoc: failed to add pmdown_time sysfs:%d\n", ret); /* create the pcm */ ret = soc_new_pcm(rtd, num); if (ret < 0) { - printk(KERN_ERR "asoc: can't create pcm %s\n", dai_link->stream_name); + pr_err("asoc: can't create pcm %s :%d\n", + dai_link->stream_name, ret); return ret; } @@ -1304,7 +1308,7 @@ static int soc_register_ac97_dai_link(struct snd_soc_pcm_runtime *rtd) ret = soc_ac97_dev_register(rtd->codec); if (ret < 0) { - printk(KERN_ERR "asoc: AC97 device register failed\n"); + pr_err("asoc: AC97 device register failed:%d\n", ret); return ret; } @@ -1444,8 +1448,8 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) ret = snd_card_create(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, card->owner, 0, &card->snd_card); if (ret < 0) { - printk(KERN_ERR "asoc: can't create sound card for card %s\n", - card->name); + pr_err("asoc: can't create sound card for card %s: %d\n", + card->name, ret); mutex_unlock(&card->mutex); return; } @@ -1563,7 +1567,8 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) ret = snd_card_register(card->snd_card); if (ret < 0) { - printk(KERN_ERR "asoc: failed to register soundcard for %s\n", card->name); + pr_err("asoc: failed to register soundcard for %s: %d\n", + card->name, ret); goto probe_aux_dev_err; } @@ -1572,7 +1577,8 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) for (i = 0; i < card->num_rtd; i++) { ret = soc_register_ac97_dai_link(&card->rtd[i]); if (ret < 0) { - printk(KERN_ERR "asoc: failed to register AC97 %s\n", card->name); + pr_err("asoc: failed to register AC97 %s: %d\n", + card->name, ret); while (--i >= 0) soc_unregister_ac97_dai_link(card->rtd[i].codec); goto probe_aux_dev_err; @@ -3070,7 +3076,7 @@ static inline char *fmt_multiple_name(struct device *dev, struct snd_soc_dai_driver *dai_drv) { if (dai_drv->name == NULL) { - printk(KERN_ERR "asoc: error - multiple DAI %s registered with no name\n", + pr_err("asoc: error - multiple DAI %s registered with no name\n", dev_name(dev)); return NULL; } @@ -3542,8 +3548,7 @@ static int __init snd_soc_init(void) #ifdef CONFIG_DEBUG_FS snd_soc_debugfs_root = debugfs_create_dir("asoc", NULL); if (IS_ERR(snd_soc_debugfs_root) || !snd_soc_debugfs_root) { - printk(KERN_WARNING - "ASoC: Failed to create debugfs directory\n"); + pr_warn("ASoC: Failed to create debugfs directory\n"); snd_soc_debugfs_root = NULL; } -- 1.7.1

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Re: [alsa-devel] tea575x-tuner improvements & use in maxiradio
by Ondrej Zary 17 Feb '12

17 Feb '12

On Wednesday 08 February 2012 08:29:25 Hans Verkuil wrote: > On Tuesday, February 07, 2012 23:20:19 Ondrej Zary wrote: > > On Sunday 05 February 2012 14:17:05 Hans Verkuil wrote: > > > These patches improve the tea575x-tuner module to make it up to date > > > with the latest V4L2 frameworks. > > > > > > The maxiradio driver has also been converted to use the tea575x-tuner > > > and I've used that card to test it. > > > > > > Unfortunately, this card can't read the data pin, so the new hardware > > > seek functionality has been tested only partially (yes, it seeks, but > > > when it finds a channel I can't read back the frequency). > > > > > > Ondrej, are you able to test these patches for the sound cards that use > > > this tea575x tuner? > > > > > > Note that these two patches rely on other work that I did and that > > > hasn't been merged yet. So it is best to pull from my git tree: > > > > > > http://git.linuxtv.org/hverkuil/media_tree.git/shortlog/refs/heads/radi > > >o-pc i2 > > > > > > You can use the v4l-utils repository > > > (http://git.linuxtv.org/v4l-utils.git) to test the drivers: the > > > v4l2-compliance test should succeed and with v4l2-ctl you can test the > > > hardware seek: > > > > > > To seek down: > > > > > > v4l2-ctl -d /dev/radio0 --freq-seek=dir=0 > > > > > > To seek up: > > > > > > v4l2-ctl -d /dev/radio0 --freq-seek=dir=1 > > > > > > To do the compliance test: > > > > > > v4l2-compliance -r /dev/radio0 > > > > It seems to work (tested with SF64-PCR - snd_fm801) but the seek is > > severely broken. Reading the frequency immediately after seek does not > > work, it always returns the old value (haven't found a delay that works). > > Reading it later (copied back snd_tea575x_get_freq function) works. The > > chip seeks randomly up or down, ignoring UP/DOWN flag and often stops at > > wrong place (only noise) or even outside the FM range. > > > > So I strongly suggest not to enable this (mis-)feature. The HW seems to > > be completely broken (unless there's some weird bug in the code). > > Well, it seemed like a good idea at the time :-) I'll remove this > 'feature', it's really not worth our time to try and make this work for > these old cards. I fixed it somehow. Now it works most of the time. The important things: - a delay must be present after search start and before first register read or the seek does weird things - when the search stops, the new frequency is not available immediately, we must wait until it appears in the register (fortunately, we can clear the frequency bits when starting the search as it starts at the frequency currently set, not from the value written) - sometimes, seek remains on the current frequency (or moves only a little), so repeat it until it moves by at least 50 kHz --- a/sound/i2c/other/tea575x-tuner.c +++ b/sound/i2c/other/tea575x-tuner.c @@ -89,7 +89,7 @@ static void snd_tea575x_write(struct snd_tea575x *tea, unsigned int val) tea->ops->set_pins(tea, 0); } -static unsigned int snd_tea575x_read(struct snd_tea575x *tea) +static u32 snd_tea575x_read(struct snd_tea575x *tea) { u16 l, rdata; u32 data = 0; @@ -120,6 +120,27 @@ static unsigned int snd_tea575x_read(struct snd_tea575x *tea) return data; } +static void snd_tea575x_get_freq(struct snd_tea575x *tea) +{ + u32 freq = snd_tea575x_read(tea) & TEA575X_BIT_FREQ_MASK; + + if (freq == 0) { + tea->freq = 0; + return; + } + + /* freq *= 12.5 */ + freq *= 125; + freq /= 10; + /* crystal fixup */ + if (tea->tea5759) + freq += TEA575X_FMIF; + else + freq -= TEA575X_FMIF; + + tea->freq = clamp(freq * 16, FREQ_LO, FREQ_HI); /* from kHz */ +} + static void snd_tea575x_set_freq(struct snd_tea575x *tea) { u32 freq = tea->freq; @@ -203,6 +224,8 @@ static int vidioc_g_frequency(struct file *file, void *priv, if (f->tuner != 0) return -EINVAL; f->type = V4L2_TUNER_RADIO; + if (!tea->cannot_read_data) + snd_tea575x_get_freq(tea); f->frequency = tea->freq; return 0; } @@ -225,36 +248,50 @@ static int vidioc_s_hw_freq_seek(struct file *file, void *fh, struct v4l2_hw_freq_seek *a) { struct snd_tea575x *tea = video_drvdata(file); + int i, old_freq; + unsigned long timeout; if (tea->cannot_read_data) return -ENOTTY; + + snd_tea575x_get_freq(tea); + old_freq = tea->freq; + /* clear the frequency, HW will fill it in */ + tea->val &= ~TEA575X_BIT_FREQ_MASK; tea->val |= TEA575X_BIT_SEARCH; - tea->val &= ~TEA575X_BIT_UPDOWN; if (a->seek_upward) tea->val |= TEA575X_BIT_UPDOWN; + else + tea->val &= ~TEA575X_BIT_UPDOWN; snd_tea575x_write(tea, tea->val); + timeout = jiffies + msecs_to_jiffies(10000); for (;;) { - unsigned val = snd_tea575x_read(tea); - - if (!(val & TEA575X_BIT_SEARCH)) { - /* Found a frequency */ - val &= TEA575X_BIT_FREQ_MASK; - val = (val * 10) / 125; - if (tea->tea5759) - val += TEA575X_FMIF; - else - val -= TEA575X_FMIF; - tea->freq = clamp(val * 16, FREQ_LO, FREQ_HI); - return 0; - } + if (time_after(jiffies, timeout)) + break; if (schedule_timeout_interruptible(msecs_to_jiffies(10))) { /* some signal arrived, stop search */ tea->val &= ~TEA575X_BIT_SEARCH; snd_tea575x_write(tea, tea->val); return -ERESTARTSYS; } + if (!(snd_tea575x_read(tea) & TEA575X_BIT_SEARCH)) { + /* Found a frequency, wait until it can be read */ + for (i = 0; i < 100; i++) { + msleep(10); + snd_tea575x_get_freq(tea); + if (tea->freq != 0) /* available */ + break; + } + /* if we moved by less than 50 kHz, continue seeking */ + if (abs(old_freq - tea->freq) < 16 * 500) { + snd_tea575x_write(tea, tea->val); + continue; + } + tea->val &= ~TEA575X_BIT_SEARCH; + return 0; + } } - return 0; + return -ETIMEDOUT; } static int tea575x_s_ctrl(struct v4l2_ctrl *ctrl) @@ -318,7 +355,7 @@ int snd_tea575x_init(struct snd_tea575x *tea) return -ENODEV; } - tea->val = TEA575X_BIT_BAND_FM | TEA575X_BIT_SEARCH_10_40; + tea->val = TEA575X_BIT_BAND_FM | TEA575X_BIT_SEARCH_5_28; tea->freq = 90500 * 16; /* 90.5Mhz default */ snd_tea575x_set_freq(tea); -- Ondrej Zary

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[alsa-devel] [PATCH] ASoC: wm8996: Make sure we bounce /RESET to reset
by Mark Brown 17 Feb '12

17 Feb '12

While it matches the current code only bringing the device out of reset isn't actually doing what the function says so make sure we set the GPIO high before we pull it low. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index e3e3a37..03d191a 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -1713,6 +1713,7 @@ static bool wm8996_volatile_register(struct device *dev, unsigned int reg) static int wm8996_reset(struct wm8996_priv *wm8996) { if (wm8996->pdata.ldo_ena > 0) { + gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 1); gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); return 0; } else { -- 1.7.9

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[alsa-devel] Bluetooth: alsa says connection refused?
by Fabrice DELENTE 17 Feb '12

17 Feb '12

Hello. I have a bluetooth headset that I have successfully used with a cell phone and an Android tablet, and that I could use in Ubuntu on a friend's machine. I'd like to use it with mplayer on my laptop (under Slackware 13.37). I have the BT address, and the test-audio script in bluez-4.98/test works (at least, I hear two beeps in the headphone, and hcitool con shows # hcitool con Connections: < ACL 00:18:91:D0:7A:24 handle 42 state 1 lm MASTER AUTH ENCRYPT so I guess it is possible to connnect to the device. However, if I try hcitool cc $HEADSET, the connection is closed immediately...) I created an ~/.asoundrc with: pcm.btheadset { type plug slave { pcm { type bluetooth device "00:18:91:D0:7A:24" profile "auto" } } hint { show on description "BlueTooth Halterrego H.EAR 01 Headset" } } ctl.btheadset { type bluetooth } but when I try to use it with mplayer: # mplayer -ao alsa:device=btheadset file.mp3 I get bt_audio_service_open: connect() failed: Connection refused (111) [AO_ALSA] Playback open error: Connection refused Failed to initialize audio driver 'alsa:device=btheadset' Could not open/initialize audio device -> no sound. Audio: no sound Video: no video Exiting... (End of file) What did I miss? (I recompiled bluez with support for alsa and alsa with support for bluetooth) Thanks! -- F. Delente

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