Alsa-devel March 2008

alsa-devel@alsa-project.org

113 participants
166 discussions

[alsa-devel] mm functions vs. normal IO
by Reiner 04 Mar '08

04 Mar '08

Hello, I'am new for using ALSA API. I was astonished that a simple wav playback example does not exist. But after reading the pcm examples I have managed to make one (if I will finish it I can post it so it can be added to the examples) Now my question: What should be used: snd_pcm_mmap_commit <http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m___direct.html#g…> (I think snd_pcm_mmap_writei <http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m___direct.html#g…> is the same with an additional copy behind ?) or snd_pcm_writei <http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m.html#gf13067c0e…> I think mmap will have better performance, but will it work with all soundcards? Or should it be in a config what to use? I'am also looking for a function to get the number of played frames. I have played around with "snd_pcm_status" but it seems not to have a counter for this. By the way is there a detailed tutorial (introduction) for ALSA available? The online help is a little bit too short for newbies. ;-) Reiner

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[alsa-devel] [PATCH] ALC288 - Add NEC S970 to the quirk table
by Pascal Terjan 04 Mar '08

04 Mar '08

Hi, NEC S970 has no sound in the internal speakers when autodetection is used. With targa-dig model, there is sound in the speakers and it gets correctly muted when pluging headphones.

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[alsa-devel] Question about snd_pcm_update_hw_ptr_post()
by Timur Tabi 04 Mar '08

04 Mar '08

Can someone explain to me this snippet from snd_pcm_update_hw_ptr_post(): if (avail >= runtime->stop_threshold) { if (substream->runtime->status->state == SNDRV_PCM_STATE_DRAINING) snd_pcm_drain_done(substream); else xrun(substream); return -EPIPE; } During playback, avail = snd_pcm_playback_avail(). Does this mean that avail is number of frames available in the playback DMA buffer for new data? If so, why is avail >= runtime->stop_threshold considered an underrun condition? Does ALSA try to keep the amount of available data to a minimum? If so, how can I adjust runtime->stop_threshold? I'd like to see what happens if I make stop_threshold equal to the size of my DMA buffer. -- Timur Tabi Linux kernel developer at Freescale

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[alsa-devel] [PATCH 4/4] Convert non-SoC PXA2xx AC97 driver to clock API
by Mark Brown 04 Mar '08

04 Mar '08

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Acked-by: Takashi Iwai <tiwai(a)suse.de> Cc: alsa-devel(a)alsa-project.org --- sound/arm/pxa2xx-ac97.c | 48 +++++++++++++++++++++++++++++++++++++++------- 1 files changed, 40 insertions(+), 8 deletions(-) diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 5d86e68..fd7bc86 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -16,6 +16,7 @@ #include <linux/platform_device.h> #include <linux/interrupt.h> #include <linux/wait.h> +#include <linux/clk.h> #include <linux/delay.h> #include <sound/core.h> @@ -35,6 +36,10 @@ static DEFINE_MUTEX(car_mutex); static DECLARE_WAIT_QUEUE_HEAD(gsr_wq); static volatile long gsr_bits; +static struct clk *ac97_clk; +#ifdef CONFIG_PXA27x +static struct clk *ac97conf_clk; +#endif /* * Beware PXA27x bugs: @@ -112,9 +117,9 @@ static void pxa2xx_ac97_reset(struct snd_ac97 *ac97) gsr_bits = 0; #ifdef CONFIG_PXA27x /* PXA27x Developers Manual section 13.5.2.2.1 */ - pxa_set_cken(CKEN_AC97CONF, 1); + clk_enable(ac97conf_clk); udelay(5); - pxa_set_cken(CKEN_AC97CONF, 0); + clk_disable(ac97conf_clk); GCR = GCR_COLD_RST; udelay(50); #else @@ -259,7 +264,7 @@ static int pxa2xx_ac97_do_suspend(struct snd_card *card, pm_message_t state) if (platform_ops && platform_ops->suspend) platform_ops->suspend(platform_ops->priv); GCR |= GCR_ACLINK_OFF; - pxa_set_cken(CKEN_AC97, 0); + clk_disable(ac97_clk); return 0; } @@ -268,7 +273,7 @@ static int pxa2xx_ac97_do_resume(struct snd_card *card) { pxa2xx_audio_ops_t *platform_ops = card->dev->platform_data; - pxa_set_cken(CKEN_AC97, 1); + clk_enable(ac97_clk); if (platform_ops && platform_ops->resume) platform_ops->resume(platform_ops->priv); snd_ac97_resume(pxa2xx_ac97_ac97); @@ -335,8 +340,21 @@ static int __devinit pxa2xx_ac97_probe(struct platform_device *dev) #ifdef CONFIG_PXA27x /* Use GPIO 113 as AC97 Reset on Bulverde */ pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); + ac97conf_clk = clk_get(&dev->dev, "AC97CONFCLK"); + if (IS_ERR(ac97conf_clk)) { + ret = PTR_ERR(ac97conf_clk); + ac97conf_clk = NULL; + goto err; + } #endif - pxa_set_cken(CKEN_AC97, 1); + + ac97_clk = clk_get(&dev->dev, "AC97CLK"); + if (IS_ERR(ac97_clk)) { + ret = PTR_ERR(ac97_clk); + ac97_clk = NULL; + goto err; + } + clk_enable(ac97_clk); ret = snd_ac97_bus(card, 0, &pxa2xx_ac97_ops, NULL, &ac97_bus); if (ret) @@ -361,11 +379,19 @@ static int __devinit pxa2xx_ac97_probe(struct platform_device *dev) err: if (card) snd_card_free(card); - if (CKEN & (1 << CKEN_AC97)) { + if (ac97_clk) { GCR |= GCR_ACLINK_OFF; free_irq(IRQ_AC97, NULL); - pxa_set_cken(CKEN_AC97, 0); + clk_disable(ac97_clk); + clk_put(ac97_clk); + ac97_clk = NULL; + } +#ifdef CONFIG_PXA27x + if (ac97conf_clk) { + clk_put(ac97conf_clk); + ac97conf_clk = NULL; } +#endif return ret; } @@ -378,7 +404,13 @@ static int __devexit pxa2xx_ac97_remove(struct platform_device *dev) platform_set_drvdata(dev, NULL); GCR |= GCR_ACLINK_OFF; free_irq(IRQ_AC97, NULL); - pxa_set_cken(CKEN_AC97, 0); + clk_disable(ac97_clk); + clk_put(ac97_clk); + ac97_clk = NULL; +#ifdef CONFIG_PXA27x + clk_put(ac97conf_clk); + ac97conf_clk = NULL; +#endif } return 0; -- 1.5.4.3

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[alsa-devel] [PATCH] Add Invert Switch for ROUT2
by Mark Brown 03 Mar '08

03 Mar '08

From: Graeme Gregory <graeme(a)openmoko.org> GTA02 device has a speaker between LOUT2 & ROUT2 and in this mode ROUT2 needs to be inverted. This patch adds a mixer control for this. Signed-off-by: Graeme Gregory <graeme(a)openmoko.org> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8753.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 0288275..76a5c7b 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -198,6 +198,7 @@ static const char *wm8753_mic_sel[] = {"Mic 1", "Mic 2", "Mic 3"}; static const char *wm8753_dai_mode[] = {"DAI 0", "DAI 1", "DAI 2", "DAI 3"}; static const char *wm8753_dat_sel[] = {"Stereo", "Left ADC", "Right ADC", "Channel Swap"}; +static const char *wm8753_rout2_phase[] = {"Non Inverted", "Inverted"}; static const struct soc_enum wm8753_enum[] = { SOC_ENUM_SINGLE(WM8753_BASS, 7, 2, wm8753_base), @@ -228,6 +229,7 @@ SOC_ENUM_SINGLE(WM8753_ADC, 4, 2, wm8753_adc_filter), SOC_ENUM_SINGLE(WM8753_MICBIAS, 6, 3, wm8753_mic_sel), SOC_ENUM_SINGLE(WM8753_IOCTL, 2, 4, wm8753_dai_mode), SOC_ENUM_SINGLE(WM8753_ADC, 7, 4, wm8753_dat_sel), +SOC_ENUM_SINGLE(WM8753_OUTCTL, 2, 2, wm8753_rout2_phase), }; @@ -330,6 +332,7 @@ SOC_SINGLE("Mic1 Capture Volume", WM8753_INCTL1, 5, 3, 0), SOC_ENUM_EXT("DAI Mode", wm8753_enum[26], wm8753_get_dai, wm8753_set_dai), SOC_ENUM("ADC Data Select", wm8753_enum[27]), +SOC_ENUM("ROUT2 Phase", wm8753_enum[28]), }; /* add non dapm controls */ -- 1.5.4.3

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[alsa-devel] ALSA C++ API updated for 1.0.16
by Lasse Kärkkäinen 01 Mar '08

01 Mar '08

Here's an updated version of the C++ wrapper that I posted earlier in September 2007. I would like this to be included in the ALSA distribution after some peer review. This wrapper has been used in UltraStar NG (a game shipped by Gentoo and others) since September, with no known issues other than that it is incomplete (very PCM-centric). As a reminder, the API closely follows the C API, and also allows mixing direct C API calls with itself transparently.

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Alsa-devel March 2008