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[alsa-devel] [PATCH][ASoC v2] Replace snd_soc_free_codec with snd_soc_unregister_codec
by Timur Tabi 24 Apr '08

24 Apr '08

The functions snd_soc_register_codec() and snd_soc_free_codec() are companion functions, but snd_soc_free_codec() releases resources (via kfree) that were not allocated by snd_soc_register_codec(). This means that the caller of snd_soc_register_codec() has to use a specific method to allocate these resources. So we replace snd_soc_free_codec() with snd_soc_unregister_codec(). This new function only unregisters the codec but does not free the memory resources. The caller can now allocate and free the memory resources as it sees fit. Also updated all callers of snd_soc_free_codec() to call snd_soc_unregister_codec() instead and free the resources themselves. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- include/sound/soc-codec.h | 2 +- sound/soc/codecs/wm8350.c | 8 ++++++-- sound/soc/codecs/wm8731.c | 8 ++++++-- sound/soc/codecs/wm8750.c | 8 ++++++-- sound/soc/codecs/wm8753.c | 8 ++++++-- sound/soc/codecs/wm9712.c | 8 ++++++-- sound/soc/codecs/wm9713.c | 8 ++++++-- sound/soc/soc-core.c | 11 ++++------- 8 files changed, 41 insertions(+), 20 deletions(-) diff --git a/include/sound/soc-codec.h b/include/sound/soc-codec.h index 3cf4819..6f132ba 100644 --- a/include/sound/soc-codec.h +++ b/include/sound/soc-codec.h @@ -160,7 +160,7 @@ struct snd_soc_codec *snd_soc_new_codec( int snd_soc_register_codec(struct snd_soc_codec *codec, struct device *dev); -void snd_soc_free_codec(struct snd_soc_codec *codec); +void snd_soc_deregister_codec(struct snd_soc_codec *codec); int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, struct snd_ac97_bus_ops *ops, struct snd_card *card, diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 3195d13..4ea8ced 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1436,7 +1436,9 @@ static int wm8350_codec_probe(struct platform_device *pdev) codec_err: kfree(wm8350); prv_err: - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return ret; } @@ -1446,7 +1448,9 @@ static int wm8350_codec_remove(struct platform_device *pdev) struct wm8350_data *wm8350 = codec->private_data; snd_soc_unregister_codec_dai(wm8350->dai); - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); kfree(wm8350); return 0; } diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 0fe8766..2195c77 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -595,7 +595,9 @@ static int wm8731_codec_probe(struct platform_device *pdev) codec_err: kfree(wm8731); wm8731_err: - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return ret; } @@ -606,7 +608,9 @@ static int wm8731_codec_remove(struct platform_device *pdev) snd_soc_unregister_codec_dai(wm8731->dai); kfree(wm8731); - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return 0; } diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 1b0296f..23e9126 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -898,7 +898,9 @@ static int wm8750_codec_probe(struct platform_device *pdev) codec_err: kfree(wm8750); wm8750_err: - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return ret; } @@ -909,7 +911,9 @@ static int wm8750_codec_remove(struct platform_device *pdev) snd_soc_unregister_codec_dai(wm8750->dai); kfree(wm8750); - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return 0; } static struct platform_driver wm8750_codec_driver = { diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 0d795f4..1cb2d64 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1584,7 +1584,9 @@ voice_err: codec_err: kfree(codec->private_data); err: - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return ret; } @@ -1603,7 +1605,9 @@ static int wm8753_codec_remove(struct platform_device *pdev) /* free codec */ kfree(codec->private_data); - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return 0; } diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 16b6a18..7014190 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -715,7 +715,9 @@ aux_dai_err: codec_err: kfree(wm9712); priv_err: - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return ret; } @@ -728,7 +730,9 @@ static int wm9712_codec_remove(struct platform_device *pdev) snd_soc_unregister_codec_dai(wm9712->aux_dai); kfree(codec->private_data); - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return 0; } diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 5e99bbe..d9ded51 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1154,7 +1154,9 @@ aux_dai_err: codec_err: kfree(wm9713); priv_err: - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return ret; } @@ -1168,7 +1170,9 @@ static int wm9713_codec_remove(struct platform_device *pdev) snd_soc_unregister_codec_dai(wm9713->voice_dai); kfree(codec->private_data); - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return 0; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 939b543..7dc6d68 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1856,21 +1856,18 @@ int snd_soc_register_codec(struct snd_soc_codec *codec, struct device *dev) EXPORT_SYMBOL_GPL(snd_soc_register_codec); /** - * snd_soc_free_codec - unregister and free codec. + * snd_soc_unregister_codec - unregister a codec. * @codec: codec driver * - * Unregisters a codec driver with the core and frees all its resources. + * Unregisters a codec driver with the core. */ -void snd_soc_free_codec(struct snd_soc_codec *codec) +void snd_soc_unregister_codec(struct snd_soc_codec *codec) { mutex_lock(&client_mutex); list_del(&codec->list); mutex_unlock(&client_mutex); - if (codec->reg_cache) - kfree(codec->reg_cache); - kfree(codec); } -EXPORT_SYMBOL_GPL(snd_soc_free_codec); +EXPORT_SYMBOL_GPL(snd_soc_unregister_codec); /** * snd_soc_register_platform_dai - registers a platform DAI. -- 1.5.5.1

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Re: [alsa-devel] how to disable mmap and allow .copy function to be called
by Harsha priya gupta 24 Apr '08

24 Apr '08

Thank you. I tried that but on starting a playback it gives me an error that the device is in use and does not playback. Should i be enabling anything else for my .copy function to be invoked and successful playback to happen? On Thu, Apr 24, 2008 at 10:53 AM, Eliot Blennerhassett < linux(a)audioscience.com> wrote: > Harsha priya gupta wrote: > > Hi, > > > > Can anyone help me - I need to modify an existing driver such that a new > > .copy function is invoked. how to make sure that mmap is diabled? > > > > Remove SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID > > from the struct snd_pcm_hardware.info initialisation. > > -- > Eliot > > > > > -- -Harsha

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[alsa-devel] how to disable mmap and allow .copy function to be called
by Harsha priya gupta 24 Apr '08

24 Apr '08

Hi, Can anyone help me - I need to modify an existing driver such that a new .copy function is invoked. how to make sure that mmap is diabled? Thanks, Harsha

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[alsa-devel] snd_pcm_rewind() weirdness
by Lennart Poettering 24 Apr '08

24 Apr '08

Hey! On ALSA 1.0.16 with the HDA driver I am experiencing some weird things with snd_pcm_rewind() which I think are the result of a kernel bug, but I am not sure. Here's what I do to trigger this: I play some PCM. Eventually I want to rewrite what I already have written. I call snd_pcm_hwsync() and then snd_pcm_avail_update() to determine how "empty" the playback buffer is. I subtract that value from the total buffer size. The resulting value should be something resembling the number of frames that I might be able to rewind. I then subtract enough frames to keep away from the playback pointer far enough (50ms, the playback buffer is 370ms, period is 185ms). Now I pass that value to snd_pcm_rewind() and it returns something that is higher than what I passed in. Which I find quite surprising. I had expected that the returned value could be equal or smaller than what I passed in, but never higher! So, my interpretation of the API was apparently wrong. It looks like on HDA the snd_pcm_rewind() parameter is apparently rounded *up* to the next period boundary (which in this case was equivalent to the *entire* buffer)? Of course, rewinding that far immediately caused an XRUN. Which is why I think rounding up in this cases is a bad idea and the API shouldn't work this way. Rounding down is OK, rounding up is call for trouble. To tape over this I added some code to immediately call snd_pcm_forward() with the difference if I noticed that the device forwarded more than I requested. This value also got rounded up -- to twice the period size. So with this backward+forward operation I ended up at a frame index that is farther ahead than where I originally came from! So, what's the expected behaviour of snd_pcm_rewind()? Always round up? What's the rationale behind this? If this is not a bug but expected behaviour, could we at least add something for passing the direction to round to? What I am working on is the "glitch-free" playback model in PulseAudio (http://0pointer.de/blog/projects/pulse-glitch-free.html). To do this I am writing blocks smaller than the period size to the device via the mmap interface all the time. Given that on HDA _rewind() is apparently not granular to the frame size, is writing via mmap in small blocks like that OK? If it's not, how can I learn about this in advance? Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[alsa-devel] [RFC/PATCH 0/1] ALSA SoC support for OMAP
by Jarkko Nikula 24 Apr '08

24 Apr '08

Hi Here is the second version of ASoC support for OMAP. Now based to dev branch of git://opensource.wolfsonmicro.com/linux-2.6-asoc but applies to linux-omap by modifying Kconfig and Makefile in sound/soc/. Code hasn't changed a very much. Some minor clean-ups etc and support for OMAP2430 and 3430 is not included until we have updated arch/arm/plat-omap/mcbsp.c module in place supporting them (work in progress). This have the same issues than first version [1] but TODO list is different: - Pin multiplexing: This belongs to .../plat-omap/mcbsp.c, not here - Multiformat & multilink McBSP DAI: ASoC v2 - omap_mcbsp_start & _stop API change: Probably with or after updated mcbsp.c Note that include/asm-arm/arch-omap/control.h is missing from vanilla kernel at the moment but that's queued for 2.6.26. -- Jarkko 1. http://marc.info/?l=linux-omap&m=120643770810683&w=2

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[alsa-devel] [PATCH 01/11] soc - s3c24xx-i2s - Use linux/io.h
by Mark Brown 23 Apr '08

23 Apr '08

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/s3c24xx/s3c24xx-i2s.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 0de211c..c95ebe2 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -25,6 +25,7 @@ #include <linux/delay.h> #include <linux/clk.h> #include <linux/jiffies.h> +#include <linux/io.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -32,7 +33,6 @@ #include <sound/soc.h> #include <asm/hardware.h> -#include <asm/io.h> #include <asm/arch/regs-gpio.h> #include <asm/arch/regs-clock.h> #include <asm/arch/audio.h> -- 1.5.5.1

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[alsa-devel] [PATCH 1/5] soc - Support PXA3xx AC97
by Mark Brown 23 Apr '08

23 Apr '08

The PXA3xx does not support the use of interrupts during reset and access to the GPIO status requires similar handling to that for PXA27x. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/pxa/pxa2xx-ac97.c | 29 +++++++++++++++++++++++++++-- 1 files changed, 27 insertions(+), 2 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 815c153..95ae509 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -55,7 +55,7 @@ static unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97, mutex_lock(&car_mutex); /* set up primary or secondary codec/modem space */ -#ifdef CONFIG_PXA27x +#if defined(CONFIG_PXA27x) || defined(CONFIG_PXA3xx) reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE; #else if (reg == AC97_GPIO_STATUS) @@ -105,7 +105,7 @@ static void pxa2xx_ac97_write(struct snd_ac97 *ac97, unsigned short reg, mutex_lock(&car_mutex); /* set up primary or secondary codec/modem space */ -#ifdef CONFIG_PXA27x +#if defined(CONFIG_PXA27x) || defined(CONFIG_PXA3xx) reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE; #else if (reg == AC97_GPIO_STATUS) @@ -128,6 +128,9 @@ static void pxa2xx_ac97_write(struct snd_ac97 *ac97, unsigned short reg, static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97) { +#ifdef CONFIG_PXA3xx + int timeout = 100; +#endif gsr_bits = 0; #ifdef CONFIG_PXA27x @@ -138,6 +141,11 @@ static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97) GCR |= GCR_WARM_RST; pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); udelay(500); +#elif defined(CONFIG_PXA3xx) + /* Can't use interrupts */ + GCR |= GCR_WARM_RST; + while (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)) && timeout--) + mdelay(1); #else GCR |= GCR_WARM_RST | GCR_PRIRDY_IEN | GCR_SECRDY_IEN; wait_event_timeout(gsr_wq, gsr_bits & (GSR_PCR | GSR_SCR), 1); @@ -153,6 +161,16 @@ static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97) static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97) { +#ifdef CONFIG_PXA3xx + int timeout = 1000; + + /* Hold CLKBPB for 100us */ + GCR = 0; + GCR = GCR_CLKBPB; + udelay(100); + GCR = 0; +#endif + GCR &= GCR_COLD_RST; /* clear everything but nCRST */ GCR &= ~GCR_COLD_RST; /* then assert nCRST */ @@ -164,6 +182,13 @@ static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97) pxa_set_cken(CKEN_AC97CONF, 0); GCR = GCR_COLD_RST; udelay(50); +#elif defined(CONFIG_PXA3xx) + /* Can't use interrupts on PXA3xx */ + GCR &= ~(GCR_PRIRDY_IEN|GCR_SECRDY_IEN); + + GCR = GCR_WARM_RST | GCR_COLD_RST; + while (!(GSR & (GSR_PCR | GSR_SCR)) && timeout--) + mdelay(10); #else GCR = GCR_COLD_RST; GCR |= GCR_CDONE_IE|GCR_SDONE_IE; -- 1.5.5

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[alsa-devel] Detecting when user is calling himself
by Mathieu Monney 23 Apr '08

23 Apr '08

Hi all I'm trying to figure out how to detect if the user is trying to call himself because it is making some weird unwanted things on my UI when user do this. Any idea? Thank you. Regards, Mathieu Monney

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[alsa-devel] [PATCH 1/3] soc - s3c24xx-i2s - Replace __FUNCTION__ with __func__
by Mark Brown 22 Apr '08

22 Apr '08

__FUNCTION__ is GCC specific and checkpatch won't let us submit with it. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/s3c24xx/s3c24xx-i2s.c | 20 ++++++++++---------- 1 files changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 6b3cbbf..7d38f55 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -89,7 +89,7 @@ static void s3c24xx_snd_txctrl(int on) u32 iiscon; u32 iismod; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON); iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); @@ -134,7 +134,7 @@ static void s3c24xx_snd_rxctrl(int on) u32 iiscon; u32 iismod; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON); iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); @@ -182,7 +182,7 @@ static int s3c24xx_snd_lrsync(void) u32 iiscon; int timeout = 50; /* 5ms */ - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); while (1) { iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); @@ -202,7 +202,7 @@ static int s3c24xx_snd_lrsync(void) */ static inline int s3c24xx_snd_is_clkmaster(void) { - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); return (readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & S3C2410_IISMOD_SLAVE) ? 0:1; } @@ -215,7 +215,7 @@ static int s3c24xx_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai, { u32 iismod; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); DBG("hw_params r: IISMOD: %lx \n", iismod); @@ -251,7 +251,7 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; u32 iismod; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) rtd->dai->cpu_dai->dma_data = &s3c24xx_i2s_pcm_stereo_out; @@ -279,7 +279,7 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd) { int ret = 0; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -321,7 +321,7 @@ static int s3c24xx_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, { u32 iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); iismod &= ~S3C2440_IISMOD_MPLL; @@ -347,7 +347,7 @@ static int s3c24xx_i2s_set_clkdiv(struct snd_soc_cpu_dai *cpu_dai, { u32 reg; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); switch (div_id) { case S3C24XX_DIV_BCLK: @@ -382,7 +382,7 @@ EXPORT_SYMBOL_GPL(s3c24xx_i2s_get_clockrate); static int s3c24xx_i2s_probe(struct platform_device *pdev) { - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); s3c24xx_i2s.regs = ioremap(S3C2410_PA_IIS, 0x100); if (s3c24xx_i2s.regs == NULL) -- 1.5.5

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[alsa-devel] envy24control can't set sample rate on Delta 1010
by John Rigg 22 Apr '08

22 Apr '08

envy24control in alsa-tools-1.0.16 cannot set sample rate on the M-Audio Delta 1010. The Actual Rate window appears to be functioning correctly, as is the "S/PDIF in" button, but none of the internal clock sample rate buttons do anything. Setting the sample rate as a jackd parameter works as normal. I tried it with 2.6.20 and 2.6.25 kernels. Behaviour was the same on both. I've made a bug report (0003915). John

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