June 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: TWL4030: Add EXT_MUTE gpio to reduce pop-noise effect
by Jorge Eduardo Candelaria 27 Jun '09

27 Jun '09

According to TRM, an external FET controlled by a 1.8V output signal can be used to reduce the pop-noise heard when the audio amplifier is switched on. The signal is controlled by a gpio pin and should be defined in the machine driver. However, the codec driver takes care of enabling and disabling this output during the headset pop attenuation sequence. If the board does not have an EXT_MUTE, then the machine driver should set the value of ext_mute_gpio to -EINVAL through ramp delay setup data, to bypass execution of the code that manages the gpio line. Also add a delay to let VMID settle in ramp up sequence. Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> --- sound/soc/codecs/twl4030.c | 23 +++++++++++++++++++++++ sound/soc/codecs/twl4030.h | 1 + 2 files changed, 24 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index df42fa2..e850b30 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -27,6 +27,7 @@ #include <linux/i2c.h> #include <linux/platform_device.h> #include <linux/i2c/twl4030.h> +#include <linux/gpio.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -620,6 +621,9 @@ static int handsfreerpga_event(struct snd_soc_dapm_widget *w, static void headset_ramp(struct snd_soc_codec *codec, int ramp) { + struct snd_soc_device *socdev = codec->socdev; + struct twl4030_setup_data *setup = socdev->codec_data; + unsigned char hs_gain, hs_pop; struct twl4030_priv *twl4030 = codec->private_data; /* Base values for ramp delay calculation: 2^19 - 2^26 */ @@ -629,6 +633,11 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp) hs_gain = twl4030_read_reg_cache(codec, TWL4030_REG_HS_GAIN_SET); hs_pop = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET); + /* Enable external mute, this dramatically reduces + * the pop-noise */ + if (setup && gpio_is_valid(setup->ext_mute_gpio)) + gpio_set_value(setup->ext_mute_gpio, 1); + if (ramp) { /* Headset ramp-up according to the TRM */ hs_pop |= TWL4030_VMID_EN; @@ -636,6 +645,9 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp) twl4030_write(codec, TWL4030_REG_HS_GAIN_SET, hs_gain); hs_pop |= TWL4030_RAMP_EN; twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop); + /* Wait ramp delay time + 1, so the VMID can settle */ + mdelay((ramp_base[(hs_pop & TWL4030_RAMP_DELAY) >> 2] / + twl4030->sysclk) + 1); } else { /* Headset ramp-down _not_ according to * the TRM, but in a way that it is working */ @@ -652,6 +664,10 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp) hs_pop &= ~TWL4030_VMID_EN; twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop); } + + /* Disable external mute */ + if (setup && gpio_is_valid(setup->ext_mute_gpio)) + gpio_set_value(setup->ext_mute_gpio, 0); } static int headsetlpga_event(struct snd_soc_dapm_widget *w, @@ -2101,6 +2117,7 @@ static int twl4030_init(struct snd_soc_device *socdev) /* Configuration for headset ramp delay from setup data */ if (setup) { unsigned char hs_pop; + unsigned int ext_mute; if (setup->sysclk) twl4030->sysclk = setup->sysclk; @@ -2111,6 +2128,12 @@ static int twl4030_init(struct snd_soc_device *socdev) hs_pop &= ~TWL4030_RAMP_DELAY; hs_pop |= (setup->ramp_delay_value << 2); twl4030_write_reg_cache(codec, TWL4030_REG_HS_POPN_SET, hs_pop); + + if (gpio_is_valid(setup->ext_mute_gpio)) { + ext_mute = setup->ext_mute_gpio; + BUG_ON(gpio_request(ext_mute, "ext_mute") < 0); + gpio_direction_output(ext_mute, 0); + } } else { twl4030->sysclk = 26000; } diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index fe5f395..23a75e2 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -274,6 +274,7 @@ extern struct snd_soc_codec_device soc_codec_dev_twl4030; struct twl4030_setup_data { unsigned int ramp_delay_value; unsigned int sysclk; + unsigned int ext_mute_gpio; }; #endif /* End of __TWL4030_AUDIO_H__ */ -- 1.6.0.4

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[alsa-devel] routing audio stream to multiple devices
by Baek Chang 26 Jun '09

26 Jun '09

I was wondering if it was possible to route a single audio stream to multiple alsa devices or plugins? For example, I am playing a song and want it to stream to both my sound card at hw:0 and a bluetooth device, an alsa plugin. Is this possible? It seems that configuring asound.conf only lets you route streams serially to a single device at a time. Any info would be greatly appreciated. Thanks in advance Baek -- Baek Chang baeksan(a)ccrma.stanford.edu 407-267-6701

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[alsa-devel] [PATCH] ASoC: Remove word "Switch" from Handsfree switch name
by Lopez Cruz, Misael 26 Jun '09

26 Jun '09

SoC dapm adds the suffix "Switch" to SND_SOC_DAPM_SWITCH controls, removing word "Switch" from HandsfreeL/HandsfreeR widget name for avoiding to duplicate it. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl4030.c | 12 ++++++------ 1 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index df42fa2..49ceb62 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1258,14 +1258,14 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { /* HandsfreeL/R */ SND_SOC_DAPM_MUX("HandsfreeL Mux", SND_SOC_NOPM, 0, 0, &twl4030_dapm_handsfreel_control), - SND_SOC_DAPM_SWITCH("HandsfreeL Switch", SND_SOC_NOPM, 0, 0, + SND_SOC_DAPM_SWITCH("HandsfreeL", SND_SOC_NOPM, 0, 0, &twl4030_dapm_handsfreelmute_control), SND_SOC_DAPM_PGA_E("HandsfreeL PGA", SND_SOC_NOPM, 0, 0, NULL, 0, handsfreelpga_event, SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_MUX("HandsfreeR Mux", SND_SOC_NOPM, 5, 0, &twl4030_dapm_handsfreer_control), - SND_SOC_DAPM_SWITCH("HandsfreeR Switch", SND_SOC_NOPM, 0, 0, + SND_SOC_DAPM_SWITCH("HandsfreeR", SND_SOC_NOPM, 0, 0, &twl4030_dapm_handsfreermute_control), SND_SOC_DAPM_PGA_E("HandsfreeR PGA", SND_SOC_NOPM, 0, 0, NULL, 0, handsfreerpga_event, @@ -1377,15 +1377,15 @@ static const struct snd_soc_dapm_route intercon[] = { {"HandsfreeL Mux", "AudioL1", "Analog L1 Playback Mixer"}, {"HandsfreeL Mux", "AudioL2", "Analog L2 Playback Mixer"}, {"HandsfreeL Mux", "AudioR2", "Analog R2 Playback Mixer"}, - {"HandsfreeL Switch", "Switch", "HandsfreeL Mux"}, - {"HandsfreeL PGA", NULL, "HandsfreeL Switch"}, + {"HandsfreeL", "Switch", "HandsfreeL Mux"}, + {"HandsfreeL PGA", NULL, "HandsfreeL"}, /* HandsfreeR */ {"HandsfreeR Mux", "Voice", "Analog Voice Playback Mixer"}, {"HandsfreeR Mux", "AudioR1", "Analog R1 Playback Mixer"}, {"HandsfreeR Mux", "AudioR2", "Analog R2 Playback Mixer"}, {"HandsfreeR Mux", "AudioL2", "Analog L2 Playback Mixer"}, - {"HandsfreeR Switch", "Switch", "HandsfreeR Mux"}, - {"HandsfreeR PGA", NULL, "HandsfreeR Switch"}, + {"HandsfreeR", "Switch", "HandsfreeR Mux"}, + {"HandsfreeR PGA", NULL, "HandsfreeR"}, /* Vibra */ {"Vibra Mux", "AudioL1", "DAC Left1"}, {"Vibra Mux", "AudioR1", "DAC Right1"}, -- 1.5.4.3

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[alsa-devel] RFC: ASoC driver for S3C24XX Simtec boards
by Ben Dooks 26 Jun '09

26 Jun '09

I'd like some comments on the following driver which is currently being developed to get ASoC working on the TLV320AIC23 based Simtec boards. My question is: 1) I'm not seeing a control to enable/disable the 'Speaker Amp', do I need to change the controls or how this is exported? 2) There are two boards with an GPIO AMP which has two gain controls. This is a TPA2001D1 which provides a gain of 6,12,18 or 23.5dB depending on the GPIO settings. [ http://focus.ti.com/docs/prod/folders/print/tpa2001d1.html ] This driver has yet to be tested on all our boards. /* sound/soc/s3c24xx/s3c24xx_simtec.c * * Copyright 2009 Simtec Electronics * * 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. */ #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/platform_device.h> #include <linux/gpio.h> #include <linux/clk.h> #include <linux/i2c.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> #include <sound/soc-dapm.h> #include <plat/audio-simtec.h> #include "s3c24xx-pcm.h" #include "s3c24xx-i2s.h" #include "../codecs/tlv320aic23.h" /* supported machines: * * Machine Connections AMP * ------- ----------- --- * BAST MIC, HPOUT, LOUT, LIN TPA2001D1 (HPOUTL,R) (gain hardwired) * VR1000 HPOUT, LIN None * VR2000 LIN, LOUT, MIC, HP LM4871 (HPOUTL,R) * DePicture LIN, LOUT, MIC, HP LM4871 (HPOUTL,R) * Anubis LIN, LOUT, MIC, HP TPA2001D1 (HPOUTL,R) */ static struct s3c24xx_audio_simtec_pdata *pdata; static struct clk *xtal_clk; /** * simtec_spk_event - handle power management event for speaker * @w: The widget for this event * @k: The sound control the event is for. * @event; The event being processed. * * Handle power management events to deal with the state of the * GPIO AMP attached to some boards. */ static int simtec_spk_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { int value = SND_SOC_DAPM_EVENT_ON(event) ? 0 : 1; gpio_set_value(pdata->amp_gpio, value); return 0; } static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_LINE("Line In", NULL), SND_SOC_DAPM_LINE("Line Out", NULL), SND_SOC_DAPM_MIC("Mic Jack", NULL), SND_SOC_DAPM_PGA_E("Speaker Amp", SND_SOC_NOPM, 0, 0, NULL, 0, simtec_spk_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), }; static const struct snd_soc_dapm_route audio_map[] = { { "Headphone Jack", NULL, "LHPOUT"}, { "Headphone Jack", NULL, "RHPOUT"}, { "Line Out", NULL, "LLOUT" }, { "Line Out", NULL, "RLOUT" }, { "LLINEIN", NULL, "Line In"}, { "RLINEIN", NULL, "Line In"}, { "MICIN", NULL, "Mic Jack"}, }; static const struct snd_soc_dapm_route amp_map[] = { { "Speaker Amp", NULL, "RHPOUT" }, { "Speaker Amp", NULL, "LHPOUT" }, }; static int spk_gain; /** * speaker_gain_get - read the speaker gain setting. * @kcontrol: The control for the speaker gain. * @ucontrol: The value that needs to be updated. * * Read the value for the AMP gain control. */ static int speaker_gain_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { ucontrol->value.integer.value[0] = spk_gain; return 0; } /** * speaker_gain_put - set the speaker gain setting. * @kcontrol: The control for the speaker gain. * @ucontrol: The value that needs to be set. * * Set the value of the speaker gain from the specified * @ucontrol setting. */ static int speaker_gain_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { int value = ucontrol->value.integer.value[0]; spk_gain = value; gpio_set_value_cansleep(pdata->amp_gain[0], value & 1); gpio_set_value_cansleep(pdata->amp_gain[1], value >> 1); return 0; } static const struct snd_kcontrol_new amp_controls[] = { SOC_SINGLE_EXT("Speaker Gain", 0, 0, 3, 0, speaker_gain_get, speaker_gain_put), }; /** * simtec_tlv320aic23_init - initialise and add controls * @codec; The codec instance to attach to. * * Attach our controls and configure the necessary codec * mappings for our sound card instance. */ static int simtec_tlv320aic23_init(struct snd_soc_codec *codec) { snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets, ARRAY_SIZE(tlv320aic23_dapm_widgets)); snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); if (pdata->amp_gpio > 0) { printk(KERN_DEBUG "%s: adding amp routes\n", __func__); snd_soc_dapm_add_routes(codec, amp_map, ARRAY_SIZE(amp_map)); snd_soc_dapm_enable_pin(codec, "Speaker Amp"); } if (pdata->amp_gain[0] > 0) { printk(KERN_DEBUG "%s: adding amp controls\n", __func__); snd_soc_add_controls(codec, amp_controls, ARRAY_SIZE(amp_controls)); } snd_soc_dapm_enable_pin(codec, "Headphone Jack"); snd_soc_dapm_enable_pin(codec, "Line In"); snd_soc_dapm_enable_pin(codec, "Line Out"); snd_soc_dapm_enable_pin(codec, "Mic Jack"); snd_soc_dapm_sync(codec); return 0; } #define CODEC_CLOCK 12000000 /** * simtec_hw_params - update hardware parameters * @substream: The audio substream instance. * @params: The parameters requested. * * Update the codec data routing and configuration settings * from the supplied data. */ static int simtec_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int ret; /* Set the CODEC as the bus clock master, I2S */ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret) { printk(KERN_ERR "%s: failed set cpu dai format\n", __func__); return ret; } /* Set the CODEC as the bus clock master */ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret) { printk(KERN_ERR "%s: failed set codec dai format\n", __func__); return ret; } ret = snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN); if (ret) { printk(KERN_ERR "%s: failed setting codec sysclk\n", __func__); return ret; } if (pdata->use_mpllin) { ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_MPLL, 0, SND_SOC_CLOCK_OUT); if (ret) { printk(KERN_ERR "%s: failed to set MPLLin as clksrc\n", __func__); return ret; } } if (pdata->output_cdclk) { int cdclk_scale; cdclk_scale = clk_get_rate(xtal_clk) / CODEC_CLOCK; cdclk_scale--; ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, cdclk_scale); } return 0; } static int simtec_hw_startup(struct snd_pcm_substream *substream) { /* call any board supplied startup code, this currently only * covers the bast/vr1000 which have a CPLD in the way of the * LRCLK */ if (pdata->startup) pdata->startup(); return 0; } static struct snd_soc_ops simtec_ops = { .startup = simtec_hw_startup, .hw_params = simtec_hw_params, }; static struct snd_soc_dai_link simtec_dai_aic23 = { .name = "tlv320aic23", .stream_name = "TLV320AIC23", .cpu_dai = &s3c24xx_i2s_dai, .codec_dai = &tlv320aic23_dai, .init = simtec_tlv320aic23_init, .ops = &simtec_ops, }; /* simtec audio machine driver */ static struct snd_soc_card snd_soc_machine_simtec_aic23 = { .name = "Simtec", .platform = &s3c24xx_soc_platform, .dai_link = &simtec_dai_aic23, .num_links = 1, }; /* simtec audio subsystem */ static struct snd_soc_device simtec_snd_devdata_aic23 = { .card = &snd_soc_machine_simtec_aic23, .codec_dev = &soc_codec_dev_tlv320aic23, }; /** * attach_gpio_amp - get and configure the necessary gpios * @dev: The device we're probing. * @pd: The platform data supplied by the board. * * If there is a GPIO based amplifier attached to the board, claim * the necessary GPIO lines for it, and set default values. */ static int attach_gpio_amp(struct device *dev, struct s3c24xx_audio_simtec_pdata *pd) { int ret; /* attach gpio amp gain (if any) */ if (pdata->amp_gain[0] > 0) { ret = gpio_request(pd->amp_gain[0], "gpio-amp-gain"); if (ret) { dev_err(dev, "cannot get amp gpio gain0\n"); return ret; } ret = gpio_request(pd->amp_gain[1], "gpio-amp-gain"); if (ret) { dev_err(dev, "cannot get amp gpio gain0\n"); gpio_free(pdata->amp_gain[0]); return ret; } gpio_direction_output(pd->amp_gain[0], 0); gpio_direction_output(pd->amp_gain[1], 0); } /* note, curently we assume GPA0 isn't valid amp */ if (pdata->amp_gpio > 0) { ret = gpio_request(pd->amp_gpio, "gpio-amp"); if (ret) { dev_err(dev, "cannot get amp gpio %d (%d)\n", pd->amp_gpio, ret); goto err_amp; } /* start with amp off */ gpio_direction_output(pd->amp_gpio, 1); } return 0; err_amp: if (pd->amp_gain[0] > 0) { gpio_free(pd->amp_gain[0]); gpio_free(pd->amp_gain[1]); } return ret; } static void detach_gpio_amp(struct s3c24xx_audio_simtec_pdata *pd) { if (pd->amp_gain[0] > 0) { gpio_free(pd->amp_gain[0]); gpio_free(pd->amp_gain[1]); } if (pd->amp_gpio > 0) gpio_free(pd->amp_gpio); } static int __devinit simtec_audio_probe(struct platform_device *pdev) { struct platform_device *snd_dev; int ret; dev_info(&pdev->dev, "probe called\n"); pdata = pdev->dev.platform_data; if (!pdata) { dev_err(&pdev->dev, "no platform data supplied\n"); return -EINVAL; } xtal_clk = clk_get(&pdev->dev, "xtal"); if (IS_ERR(xtal_clk)) { dev_err(&pdev->dev, "could not get clkout0\n"); return -EINVAL; } dev_info(&pdev->dev, "xtal rate is %ld\n", clk_get_rate(xtal_clk)); ret = attach_gpio_amp(&pdev->dev, pdata); if (ret) goto err_clk; snd_dev = platform_device_alloc("soc-audio", -1); if (!snd_dev) { dev_err(&pdev->dev, "failed to alloc soc-audio devicec\n"); ret = -ENOMEM; goto err_gpio; } platform_set_drvdata(snd_dev, &simtec_snd_devdata_aic23); simtec_snd_devdata_aic23.dev = &snd_dev->dev; ret = platform_device_add(snd_dev); if (ret) { dev_err(&pdev->dev, "failed to add soc-audio dev\n"); platform_device_put(snd_dev); goto err_clk; } platform_set_drvdata(pdev, snd_dev); return 0; err_gpio: detach_gpio_amp(pdata); err_clk: clk_put(xtal_clk); return ret; } static int __devexit simtec_audio_remove(struct platform_device *pdev) { struct platform_device *snd_dev = platform_get_drvdata(pdev); platform_device_unregister(snd_dev); detach_gpio_amp(pdata); clk_put(xtal_clk); return 0; } static struct platform_driver simtec_audio_platdrv = { .driver = { .owner = THIS_MODULE, .name = "s3c24xx-tlv320aic23", }, .probe = simtec_audio_probe, .remove = __devexit_p(simtec_audio_remove), }; MODULE_ALIAS("platform:s3c24xx-tlv320aic23"); static int __init simtec_init(void) { return platform_driver_register(&simtec_audio_platdrv); } static void __exit simtec_exit(void) { platform_driver_unregister(&simtec_audio_platdrv); } module_init(simtec_init); module_exit(simtec_exit); MODULE_AUTHOR("Ben Dooks <ben(a)simtec.co.uk>"); MODULE_DESCRIPTION("ALSA SoC Simtec Audio support"); MODULE_LICENSE("GPL");

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Re: [alsa-devel] Regarding Sound on Acer Aspire 6935G
by Emilio López 26 Jun '09

26 Jun '09

Well, it seems my last email was too heavy with the 4 files attached. I have uploaded them to drop.io, here is the link: http://drop.io/alsaonaceraspire6930 Thanks! 2009/5/23 Emilio López <buhitoescolar(a)gmail.com> > Here you are, I ran alsa-info.sh in the old alsa (with model=acer) and > saved the output of the script when side was enabled and disabled, with > volumes 0 and 100. Hope you can work out the problem with this, do not > hesitate to ask for more debug info if you need. > > Thanks for all! > > Emilio > > PS: On my previous email (cited below) I wrote "Mic sensing" incorrectly, I > wanted to write "headphones sensing". > > 2009/5/23 Takashi Iwai <tiwai(a)suse.de> > > At Sat, 23 May 2009 12:19:12 -0300, >> Emilio López wrote: >> > >> > I reinstalled 1.0.20 today and almost everything started working when I >> set >> > model=acer-aspire-4930g. Mic sensing works, my mic now records too. But >> my >> > bass does not, and I cannot control it with "Side" anymore. >> > >> > Any idea on how to fix it? The audio setup for this laptop will be fully >> > working when I get the bass to play once again. >> >> Could you check with the older version and what "Side" control changes >> in hd-audio codec registers? You can just get alsa-info.sh output >> before and after changing the "Side" control value. >> >> >> thanks, >> >> Takashi >> > >

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Re: [alsa-devel] Proposal for more reliable audio DMA.
by Mark Brown 26 Jun '09

26 Jun '09

On 24 Jun 2009, at 17:07, Jon Smirl <jonsmirl(a)gmail.com> wrote: > Trying to directly expose the ring buffer to the app seems like a good > > way to avoid a copy, but it isn't achieving that. pulse is not > decoding audio straight into the ring buffer, it decodes first and > then copies into the buffer. Pulse is using the timers to estimate the > destination for the copy. Move this copy down into the drivers, the > drivers know the correct destination for the copy. Pulse isn't just doing straight playback, a large part of what it's there for is to do software mixing. When you have multiple sources active pulse is going to be forced to do the copy as part of the mixing process so putting that bit of the buffer management in kernel won't help in the way you think it does. Part of what's going on here is that the kernel code is trying to give userspace access to the data for as long as possible.

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[alsa-devel] [PATCH] ASoC: txx9aclc: dynamically allocate dmaengine devname
by Atsushi Nemoto 25 Jun '09

25 Jun '09

Use kasprintf to allocate temporary devname string instead of a fixed size string. This fixes "FIXME" introduced on removal of BUS_ID_SIZE. Signed-off-by: Atsushi Nemoto <anemo(a)mba.ocn.ne.jp> --- sound/soc/txx9/txx9aclc.c | 10 ++++++---- 1 files changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c index 938a58a..efed64b 100644 --- a/sound/soc/txx9/txx9aclc.c +++ b/sound/soc/txx9/txx9aclc.c @@ -297,15 +297,17 @@ static int txx9aclc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, static bool filter(struct dma_chan *chan, void *param) { struct txx9aclc_dmadata *dmadata = param; - char devname[20 + 2]; /* FIXME: old BUS_ID_SIZE + 2 */ + char *devname; + bool found = false; - snprintf(devname, sizeof(devname), "%s.%d", dmadata->dma_res->name, + devname = kasprintf(GFP_KERNEL, "%s.%d", dmadata->dma_res->name, (int)dmadata->dma_res->start); if (strcmp(dev_name(chan->device->dev), devname) == 0) { chan->private = &dmadata->dma_slave; - return true; + found = true; } - return false; + kfree(devname); + return found; } static int txx9aclc_dma_init(struct txx9aclc_soc_device *dev, -- 1.5.6.5

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[alsa-devel] Reenable support for Auzentech X-Fi Prelude in snd-ctxfi module
by Kamil Kaminski 25 Jun '09

25 Jun '09

Hi guys, I noticed that latest alsa-driver-unstable does not detect my card as if PCI IDs were discarded or purposely disabled to only work on Creative cards, the earlier snapshots used to work fine. This is what I get: ctatc: Card not recognised ctxfi: Something wrong!!! SB-XFi: probe of 0000:06:01.0 failed with error -2 This is the card: 06:01.0 Multimedia audio controller: Creative Labs SB X-Fi Subsystem: Auzentech, Inc. Device 0034 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64 (1000ns min, 1250ns max) Interrupt: pin A routed to IRQ 22 Region 0: I/O ports at 8f00 [size=32] Region 1: Memory at dfc00000 (64-bit, non-prefetchable) [size=2M] Region 3: Memory at d8000000 (64-bit, non-prefetchable) [size=64M] Capabilities: [40] Power Management version 2 Flags: PMEClk- DSI+ D1+ D2+ AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [50] MSI: Mask- 64bit+ Count=1/1 Enable- Address: 0000000000000000 Data: 0000 Kernel driver in use: CTALSA Kernel modules: ctxfi I'm currently using v1.0 ctxfi alsa driver from Creative, It would be nice to have Auzentech X-Fi Prelude working in 2.6.31 kernel ;) Thanks.

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[alsa-devel] Proposal for more reliable audio DMA.
by Jon Smirl 25 Jun '09

25 Jun '09

This algorithm is fairly similar to what currently exists in ALSA with a few modifications. I'm trying to come up with guaranteed way to play glitch free audio. Does this algorithm work or could it be modified to work? The main change is to track a high resolution timer as the means of estimating where in the buffer to insert new, low latency data. This tracking is done in a timer tick interrupt. Jiffies are too coarse and not all hardware allows you to ask the current DMA position. The other change is to keep all of the buffers filled with silence if there isn't any pending data. (not doing this caused problem in my mpc5200 AC97 driver) Three buffers are used as a way to deterministically bound the DMA pointer without needing interrupts. --- Use three chained buffers. Buffer size is samples/tick (or maybe 1.5 samples/tick) Initialize buffers to silence. Set end of buffer three to automatically terminate DMA Set FIFO to minimum bus allows Fill buffer one with samples available, start playing. On timer tick call into driver.. If buffer one has finished playing move it to third position. New last buffer is set to automatically stop DMA. Return swap status to ALSA If swapped, fill buffer three with silence or pending data It is important to fill this buffer with silence if there is no pending data The size of the buffers needs to be large enough to cover worst case timer tick latency. Buffer two has to be large enough to ensure that the tick routine will notice buffer one has been played before buffer three starts. If two buffers have been completed when the tick runs, let the third buffer finish without swapping. Since you don't know where you are in the third buffer, it is unsafe to swap at this point. Playing the last buffer causes the end of buffer interrupt to happen. Callback into ALSA to alert it of the underrun and need to restart DMA. If this callback happens, make the buffers larger. Long term observation of buffer completion status in tick handler will allow accurate computation of samples/HPET unit. Record high accuracy timer (HPET) on tick. New low latency data that arrives can be inserted into the buffers dynamically. Use the high resolution timer source to estimate where to place it. Minimum FIFO ensures low latency play. Design a minimum power mode. Allows a huge FIFO to be loaded (like 128KB). Call into driver to reset to minimum latency, small FIFO mode. Advantages: 1) If ALSA goes away, hardware will stop on it's own with no noise. 2) No need to know the current position of DMA hardware. 3) Both low and high latency modes. 4) Audio interrupts only generated as error condition 5) Behavior is deterministic. Nothing is left to guess work. -- Jon Smirl jonsmirl(a)gmail.com

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Re: [alsa-devel] ASoC updates for 2.6.31
by Mark Brown 25 Jun '09

25 Jun '09

The following changes since commit 28d0325ce6e0a52f53d8af687e6427fee59004d3: Linus Torvalds (1): Linux 2.6.31-rc1 are available in the git repository at: ssh://master.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31 Peter Ujfalusi (1): MAINTAINERS: Add entry for twl4030 series soc codec driver MAINTAINERS | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-)

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