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

[alsa-devel] [PATCH] ASoC: omap-mcpdm: Add missing MODULE_ALIAS
by Peter Ujfalusi 06 Jul '12

06 Jul '12

The MODULE_ALIAS() was missing from the driver. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/omap-mcpdm.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index b5e23af..f7babb3 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -501,6 +501,7 @@ static struct platform_driver asoc_mcpdm_driver = { module_platform_driver(asoc_mcpdm_driver); +MODULE_ALIAS("platform:omap-mcpdm"); MODULE_AUTHOR("Misael Lopez Cruz <misael.lopez(a)ti.com>"); MODULE_DESCRIPTION("OMAP PDM SoC Interface"); MODULE_LICENSE("GPL"); -- 1.7.8.6

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[alsa-devel] [PATCH] ASoC: dpcm: Allow FE to be opened without valid BE routes.
by Liam Girdwood 06 Jul '12

06 Jul '12

Some userspace will open a PCM device and then configure mixers for routing before triggering. This patch allows userspace to do this sequence. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-pcm.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 7063b8f..ef22d0b 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1955,10 +1955,8 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) fe->dpcm[stream].runtime = fe_substream->runtime; if (dpcm_path_get(fe, stream, &list) <= 0) { - dev_warn(fe->dev, "asoc: %s no valid %s route\n", + dev_dbg(fe->dev, "asoc: %s no valid %s route\n", fe->dai_link->name, stream ? "capture" : "playback"); - mutex_unlock(&fe->card->mutex); - return -EINVAL; } /* calculate valid and active FE <-> BE dpcms */ -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: twl6040: fix spelling mistake
by Liam Girdwood 06 Jul '12

06 Jul '12

From: Simon Wilson <simonwilson(a)google.com> Fix spelling mistake in "High-Performance" option of twl6040 power mode. Signed-off-by: Simon Wilson <simonwilson(a)google.com> --- sound/soc/codecs/twl6040.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index a36e9fc..0ff1e70 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -553,7 +553,7 @@ static const struct snd_kcontrol_new vibrar_mux_controls = /* Headset power mode */ static const char *twl6040_power_mode_texts[] = { - "Low-Power", "High-Perfomance", + "Low-Power", "High-Performance", }; static const struct soc_enum twl6040_power_mode_enum = -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: dapm: Fix locking during codec shutdown
by Liam Girdwood 06 Jul '12

06 Jul '12

Codec shutdown performs a DAPM power sequence that might cause conflicts and/or race conditions if another stream power event is running simultaneously. Use card's dapm mutex to protect any potential race condition between them. Signed-off-by: Misael Lopez Cruz <misael.lopez(a)ti.com> Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-dapm.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0f078de..b485960 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3578,10 +3578,13 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_free); static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm) { + struct snd_soc_card *card = dapm->card; struct snd_soc_dapm_widget *w; LIST_HEAD(down_list); int powerdown = 0; + mutex_lock(&card->dapm_mutex); + list_for_each_entry(w, &dapm->card->widgets, list) { if (w->dapm != dapm) continue; @@ -3604,6 +3607,8 @@ static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm) snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_STANDBY); } + + mutex_unlock(&card->dapm_mutex); } /* -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: dapm: Make sure all dapm contexts are updated
by Liam Girdwood 06 Jul '12

06 Jul '12

Make sure we set the bias level for all DAPM contexts when changing level. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-dapm.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 912330b..0f078de 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -291,9 +291,9 @@ static int snd_soc_dapm_set_bias_level(struct snd_soc_dapm_context *dapm, if (dapm->codec->driver->set_bias_level) ret = dapm->codec->driver->set_bias_level(dapm->codec, level); - else - dapm->bias_level = level; - } + } else + dapm->bias_level = level; + if (ret != 0) goto out; -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: pcm: Clean up logging in soc_new_pcm()
by Liam Girdwood 06 Jul '12

06 Jul '12

Use dev_ style logging throughout soc_new_pcm() Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-pcm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 48fd15b..7063b8f 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2003,7 +2003,6 @@ static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream) /* create a new pcm */ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) { - struct snd_soc_codec *codec = rtd->codec; struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; @@ -2042,7 +2041,8 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) capture, &pcm); } if (ret < 0) { - printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name); + dev_err(rtd->card->dev, "can't create pcm for %s\n", + rtd->dai_link->name); return ret; } dev_dbg(rtd->card->dev, "registered pcm #%d %s\n",num, new_name); @@ -2099,14 +2099,14 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) if (platform->driver->pcm_new) { ret = platform->driver->pcm_new(rtd); if (ret < 0) { - pr_err("asoc: platform pcm constructor failed\n"); + dev_err(platform->dev, "pcm constructor failed\n"); return ret; } } pcm->private_free = platform->driver->pcm_free; out: - printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name, + dev_info(rtd->card->dev, " %s <-> %s mapping ok\n", codec_dai->name, cpu_dai->name); return ret; } -- 1.7.9.5

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[alsa-devel] [PATCH 1/2] ASoC imx-audmux: add MX31_AUDMUX_PORT7_SSI_PINS_7 define
by Torben Hohn 06 Jul '12

06 Jul '12

The MX31 Audmux also has 7 Ports. This patch adds the missing define, and makes the debugfs code iterate over that port too. Signed-off-by: Torben Hohn <torbenh(a)linutronix.de> --- sound/soc/fsl/imx-audmux.c | 2 +- sound/soc/fsl/imx-audmux.h | 1 + 2 files changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 0803274..e7c800e 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -156,7 +156,7 @@ static void __init audmux_debugfs_init(void) return; } - for (i = 0; i < MX31_AUDMUX_PORT6_SSI_PINS_6 + 1; i++) { + for (i = 0; i < MX31_AUDMUX_PORT7_SSI_PINS_7 + 1; i++) { snprintf(buf, sizeof(buf), "ssi%d", i); if (!debugfs_create_file(buf, 0444, audmux_debugfs_root, (void *)i, &audmux_debugfs_fops)) diff --git a/sound/soc/fsl/imx-audmux.h b/sound/soc/fsl/imx-audmux.h index 04ebbab..b8ff44b 100644 --- a/sound/soc/fsl/imx-audmux.h +++ b/sound/soc/fsl/imx-audmux.h @@ -14,6 +14,7 @@ #define MX31_AUDMUX_PORT4_SSI_PINS_4 3 #define MX31_AUDMUX_PORT5_SSI_PINS_5 4 #define MX31_AUDMUX_PORT6_SSI_PINS_6 5 +#define MX31_AUDMUX_PORT7_SSI_PINS_7 6 #define MX51_AUDMUX_PORT1_SSI0 0 #define MX51_AUDMUX_PORT2_SSI1 1 -- 1.7.10

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[alsa-devel] [PATCH 1/3] ASoC: dapm: Mark widgets as dirty when a route is added
by Mark Brown 06 Jul '12

06 Jul '12

If we add a new route at runtime then we'll need to recheck the connections to the affected widgets. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index ed1860c..beac5e6 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2250,6 +2250,10 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, path->connect = 0; return 0; } + + dapm_mark_dirty(wsource, "Route added"); + dapm_mark_dirty(wsink, "Route added"); + return 0; err: -- 1.7.10

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[alsa-devel] then there was "overrun" occue every time once trigger the audio recording
by Zhen Fu 06 Jul '12

06 Jul '12

Hi Clemens, Your answer is very useful to help me. Thank you very much. > case AUDIO_STREAM_CMDID_DATARXTXREQ: > len1 = get_zsp_buf_avail(&prtd->zsp_buf); > len2 = get_fw_avail(p_zsp_req, substream->stream); > len3 = get_buf_avail(&prtd->zsp_buf, substream->stream); > tsize = ((len1 < len2) ? len1 : len2); > if ((tsize == len1) && (len1 == len3) && (tsize >= \ > 2 * prtd->zsp_buf.zsp_period_bytes)) { > tsize -= prtd->zsp_buf.zsp_period_bytes; > } > This doesn't look as if tsize is always the same as the period size. > And what do the get_fw_avail and get_buf_avail functions do? The ZSP have own buffer to save data, so get_fw_avail is acquired zsp buf avial. And ZSP buffer must ensure the zsp_buf have sequenced, so do above process. -- Best Regards, Zhen Fu

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[alsa-devel] then there was "overrun" occue every time once trigger the audio recording
by Zhen Fu 06 Jul '12

06 Jul '12

Hi developer, When I use arecord command, then there was "overrun" occue every time once trigger the audio recording. I have no clue of alsa driver when and how handler xrun. In other words, how to understand function snd_pcm_update_hw_ptr0 in pcm_lib.c. Please enlighten me on this issue. Thanks a lot in advance. -- Best Regards, Zhen Fu

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