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

[alsa-devel] Play a sine wave with alsa
by Luca Longhi 11 May '12

11 May '12

I've written this code to play a sine wave but I don't think it's right. Could you tell me where I am wrong? Especially I don't know what period_time and buffer_time are. What I have to do to extend the playback time? Code: #include <alsa/asoundlib.h> #include <math.h> int main() { int err; snd_pcm_t *handle; snd_pcm_hw_params_t *params; unsigned int Fc, val; int channels=2; int size; int dir=0; unsigned int period_time= 400000; unsigned int buffer_time= 800000; snd_pcm_uframes_t frames; snd_pcm_uframes_t bufferSize; char *buffer; /* Open PCM device for playback. */ err = snd_pcm_open(&handle, "default",SND_PCM_STREAM_PLAYBACK, 0); if (err < 0) { fprintf(stderr,"unable to open pcm device: %s\n",snd_strerror(err)); exit(1); } snd_pcm_hw_params_alloca(&params); err=snd_pcm_hw_params_any(handle, params); if (err < 0) { printf("Broken configuration for playback: no configurations available: %s\n", snd_strerror(err)); return err; } err = snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED); if (err < 0) { printf("Access type not available for playback: %s\n", snd_strerror(err)); return err; } err=snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE); if (err < 0) { printf("Sample format not available for playback: %s\n", snd_strerror(err)); return err; } err = snd_pcm_hw_params_set_channels(handle, params, channels); if (err < 0) { printf("Channels count (%i) not available for playbacks: %s\n", channels, snd_strerror(err)); return err; } Fc = 5000; dir=1; err=snd_pcm_hw_params_set_rate_near(handle, params, &Fc, &dir); if (err < 0) { printf("Rate %iHz not available for playback: %s\n", Fc, snd_strerror(err)); return err; } dir=0; err = snd_pcm_hw_params_set_period_time_near(handle, params, &period_time, &dir); if (err < 0) { printf("Unable to set period time %i for playback: %s\n", period_time, snd_strerror(err)); return err; } err = snd_pcm_hw_params_set_buffer_time_near(handle, params, &buffer_time, &dir); if (err < 0) { printf("Unable to set buffer time %i for playback: %s\n", buffer_time, snd_strerror(err)); return err; } err = snd_pcm_hw_params(handle, params); if (err < 0) { printf("Unable to set hw params for playback: %s\n", snd_strerror(err)); return err; } /* Use a buffer large enough to hold one period */ snd_pcm_hw_params_get_period_size(params, &frames, 0); size = frames * 4; short buf[size]; unsigned short ampiezza; double frequenza = 500; double omega; /* Pulsazione */ int x; omega = 2 * M_PI * frequenza / 180; ampiezza = 20000; for (x = 0; x < size ; x++) { buf[x] = (short)(ampiezza * (sin (omega*x))); } err = snd_pcm_writei(handle, buf, frames); if (err == -EPIPE) { fprintf(stderr, "underrun occurred\n"); snd_pcm_prepare(handle); } else if (err < 0) { fprintf(stderr, "error from writei: %s\n", snd_strerror(err)); } else if (err != (int)frames) { fprintf(stderr, "short write, write %d frames\n", err); } snd_pcm_drain(handle); snd_pcm_close(handle); return 0; }

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[alsa-devel] [PATCH] ASoC: pcm: Staticise non-exported functions
by Mark Brown 10 May '12

10 May '12

They pollute the global namespace and cause sparse to complain. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-pcm.c | 22 ++++++++++++---------- 1 file changed, 12 insertions(+), 10 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a3ea8a7..82766af 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -632,7 +632,8 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return 0; } -int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, int cmd) +static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, + int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_platform *platform = rtd->platform; @@ -1087,7 +1088,7 @@ unwind: return err; } -void dpcm_set_fe_runtime(struct snd_pcm_substream *substream) +static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -1247,7 +1248,7 @@ static int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) return 0; } -int dpcm_fe_dai_hw_free(struct snd_pcm_substream *substream) +static int dpcm_fe_dai_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *fe = substream->private_data; int err, stream = substream->stream; @@ -1354,8 +1355,8 @@ unwind: return ret; } -int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) +static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *fe = substream->private_data; int ret, stream = substream->stream; @@ -1404,7 +1405,8 @@ static int dpcm_do_trigger(struct snd_soc_dpcm *dpcm, return ret; } -int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd) +static int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, + int cmd) { struct snd_soc_dpcm *dpcm; int ret = 0; @@ -1497,7 +1499,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd) } EXPORT_SYMBOL_GPL(dpcm_be_dai_trigger); -int dpcm_fe_dai_trigger(struct snd_pcm_substream *substream, int cmd) +static int dpcm_fe_dai_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *fe = substream->private_data; int stream = substream->stream, ret; @@ -1605,7 +1607,7 @@ static int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) return ret; } -int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) +static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *fe = substream->private_data; int stream = substream->stream, ret = 0; @@ -1935,7 +1937,7 @@ int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute) return 0; } -int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) +static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) { struct snd_soc_pcm_runtime *fe = fe_substream->private_data; struct snd_soc_dpcm *dpcm; @@ -1972,7 +1974,7 @@ int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) return ret; } -int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream) +static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream) { struct snd_soc_pcm_runtime *fe = fe_substream->private_data; struct snd_soc_dpcm *dpcm; -- 1.7.10

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[alsa-devel] [PATCH] ASoC: ux500: Don't make the CPU I2S driver user selectable
by Mark Brown 10 May '12

10 May '12

It can only be used with a machine driver so the idiomatic thing for ASoC is to select this driver from the machine driver. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/ux500/Kconfig | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/ux500/Kconfig b/sound/soc/ux500/Kconfig index 1be1f50..44cf434 100644 --- a/sound/soc/ux500/Kconfig +++ b/sound/soc/ux500/Kconfig @@ -10,7 +10,5 @@ menuconfig SND_SOC_UX500 any of the Ux500 platforms (e.g. U8500). config SND_SOC_UX500_PLAT_MSP_I2S - tristate "Platform - MSP (I2S)" + tristate depends on SND_SOC_UX500 - help - Say Y if you want to enable the Ux500 MSP I2S-driver. -- 1.7.10

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[alsa-devel] Sizeof int in Alsa
by Alan Horstmann 10 May '12

10 May '12

Hi, I have often wondered whether Alsa had an intrinsic requirement that an int would always be at least 32-bit, but been unable to find a clear statement with web searches. Looking recently at snd_pcm_hw_params_set_rate(), for example, the rate value parameter is unsigned int, which suggests to me that it would be broken with 16-bit ints; is it correct then to deduce that Alsa does not support 16-bit int platforms? Regards, Alan

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[alsa-devel] [PATCH v3] OMAPDSS: HDMI: Implement DSS driver interface for audio
by Ricardo Neri 10 May '12

10 May '12

Hi, This patch is the third version of PATCH 14 of this series: http://www.spinics.net/lists/linux-omap/msg69466.html I send this patch separately to ease review, as the rest of the patches in the series were regarded as OK by the DSS maintainer. Changes with respect to v2: *An improved locking strategy. The new strategy maintains an audio state machine. The audio state may only be transitioned if the panel is active. Audio can only be started if the previous state is AUDIO_ENABLED. Such state may only be reached if the panel is enabled. If the panel is disabled, it will also disable audio. As the panel and audio states are protected by a mutex and a spinlock, a video thread and audio thread cannot concurrently execute audio functions. BR, Ricardo Ricardo Neri (1): OMAPDSS: HDMI: Implement DSS driver interface for audio drivers/video/omap2/dss/dss.h | 8 ++ drivers/video/omap2/dss/hdmi.c | 42 ++++++++++ drivers/video/omap2/dss/hdmi_panel.c | 140 ++++++++++++++++++++++++++++++++++ 3 files changed, 190 insertions(+), 0 deletions(-) -- 1.7.5.4

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[alsa-devel] [PATCH v3] Provide interface for audio support in DSS
by Ricardo Neri 10 May '12

10 May '12

Hi, This patch is proposed as a result of a previous discussion related to include audio support in the DSS device driver (http://www.spinics.net/lists/linux-omap/msg64748.html) At the moment, this is intended to cover display technologies with audio based on CEA-861 and IEC-60958, such as HDMI and DisplayPort. However, provisions are made to make it extensible (see below). These are the changes with respect to v2: * A new enum and field is added to DSS to help implementations of the interface to keep track of the audio state. These are the changes with respect to v1: * The audio_enable and audio_start functions were split into enable/disable and start/stop respectively to improve readability. * Comments were added on whether the DSS audio functions may sleep. * In order to make the audio configuration parameters extensible, a struct omap_dss_audio is defined. This helps to easily add more config parameters in the future if/when required (e.g., speaker order configuration). BR, Ricardo Ricardo Neri (1): OMAPDSS: Provide an interface for audio support Documentation/arm/OMAP/DSS | 45 ++++++++++++++++++++++++++++++++++++++++++++ include/video/omapdss.h | 34 +++++++++++++++++++++++++++++++++ 2 files changed, 79 insertions(+), 0 deletions(-) -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: cs42l52: Staticise non-exported symbols
by Mark Brown 09 May '12

09 May '12

Makes sparse happy and avoids polluting the global namespace. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/cs42l52.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 703060d..a710941 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -919,7 +919,7 @@ static struct snd_soc_dai_ops cs42l52_ops = { .set_sysclk = cs42l52_set_sysclk, }; -struct snd_soc_dai_driver cs42l52_dai = { +static struct snd_soc_dai_driver cs42l52_dai = { .name = "cs42l52", .playback = { .stream_name = "Playback", @@ -1163,7 +1163,7 @@ static int cs42l52_remove(struct snd_soc_codec *codec) return 0; } -struct snd_soc_codec_driver soc_codec_dev_cs42l52 = { +static struct snd_soc_codec_driver soc_codec_dev_cs42l52 = { .probe = cs42l52_probe, .remove = cs42l52_remove, .suspend = cs42l52_suspend, -- 1.7.10

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[alsa-devel] [PATCH 1/2] ASoC: wm8994: Add debounce to wm8994 mic detection
by Mark Brown 09 May '12

09 May '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 27 ++++++++++++++++++++------- sound/soc/codecs/wm8994.h | 2 ++ 2 files changed, 22 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index dfc4e90..93d81a9 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3227,22 +3227,20 @@ int wm8994_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, } EXPORT_SYMBOL_GPL(wm8994_mic_detect); -static irqreturn_t wm8994_mic_irq(int irq, void *data) +static void wm8994_mic_work(struct work_struct *work) { - struct wm8994_priv *priv = data; + struct wm8994_priv *priv = container_of(work, + struct wm8994_priv, + mic_work.work); struct snd_soc_codec *codec = priv->codec; int reg; int report; -#ifndef CONFIG_SND_SOC_WM8994_MODULE - trace_snd_soc_jack_irq(dev_name(codec->dev)); -#endif - reg = snd_soc_read(codec, WM8994_INTERRUPT_RAW_STATUS_2); if (reg < 0) { dev_err(codec->dev, "Failed to read microphone status: %d\n", reg); - return IRQ_HANDLED; + return; } dev_dbg(codec->dev, "Microphone status: %x\n", reg); @@ -3284,6 +3282,20 @@ static irqreturn_t wm8994_mic_irq(int irq, void *data) snd_soc_jack_report(priv->micdet[1].jack, report, SND_JACK_HEADSET | SND_JACK_BTN_0); +} + +static irqreturn_t wm8994_mic_irq(int irq, void *data) +{ + struct wm8994_priv *priv = data; + struct snd_soc_codec *codec = priv->codec; + +#ifndef CONFIG_SND_SOC_WM8994_MODULE + trace_snd_soc_jack_irq(dev_name(codec->dev)); +#endif + + pm_wakeup_event(codec->dev, 300); + + schedule_delayed_work(&priv->mic_work, msecs_to_jiffies(250)); return IRQ_HANDLED; } @@ -3656,6 +3668,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->codec = codec; mutex_init(&wm8994->accdet_lock); + INIT_DELAYED_WORK(&wm8994->mic_work, wm8994_mic_work); for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++) init_completion(&wm8994->fll_locked[i]); diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index 91650bb..d77e06f 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -12,6 +12,7 @@ #include <sound/soc.h> #include <linux/firmware.h> #include <linux/completion.h> +#include <linux/workqueue.h> #include "wm_hubs.h" @@ -127,6 +128,7 @@ struct wm8994_priv { struct mutex accdet_lock; struct wm8994_micdet micdet[2]; + struct delayed_work mic_work; bool mic_detecting; bool jack_mic; int btn_mask; -- 1.7.10

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Re: [alsa-devel] Fast Track Ultra latency and new streaming logic.
by Takashi Iwai 09 May '12

09 May '12

At Wed, 9 May 2012 15:05:47 +0200, Felix Homann wrote: > > Am 03.05.2012 16:03 schrieb "Felix Homann" <linuxaudio(a)showlabor.de>: > > > > Is this a > > flaw in Alsa, Jack or the USB stack? > > I'm starting to think that this is a Jack issue. At least I can set rather > small period and buffer sizes in aplay without any xruns. Or a full-duplex issue? You can test the latency via alsa-lib/test/latency program. (Yes, the usage is quite undocumented. Anyone contributing the howto would be great...) Takashi

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[alsa-devel] [PATCH] ASoC: wm8994: Fix sparse warning due to use of 0 as NULL
by Mark Brown 09 May '12

09 May '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index b3b2d5f..dfc4e90 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -792,7 +792,7 @@ static void vmid_reference(struct snd_soc_codec *codec) switch (wm8994->vmid_mode) { default: - WARN_ON(0 == "Invalid VMID mode"); + WARN_ON(NULL == "Invalid VMID mode"); case WM8994_VMID_NORMAL: /* Startup bias, VMID ramp & buffer */ snd_soc_update_bits(codec, WM8994_ANTIPOP_2, -- 1.7.10

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