January 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] upsampling
by Ran Shalit 26 Jan '13

26 Jan '13

Hello, I would like to ask if I can use alsa-lib to resample (downsample or sample) input audio and playback audio. I searched for appropriate API but found none. Regards, Ran

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[alsa-devel] [patch] ASoC: dwc: fix support for more than two channels
by Dan Carpenter 26 Jan '13

26 Jan '13

There were missing break statements so everything used TWO_CHANNEL_SUPPORT. Also I added a return statement to silence a GCC warning: sound/soc/dwc/designware_i2s.c: In function ‘dw_i2s_hw_params’: sound/soc/dwc/designware_i2s.c:236:32: warning: ‘ch_reg’ may be used uninitialized in this function [-Wmaybe-uninitialized] Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- Static analysis. Untested. diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index 1aa5130..deb30d5 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -210,15 +210,19 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream, switch (config->chan_nr) { case EIGHT_CHANNEL_SUPPORT: ch_reg = 3; + break; case SIX_CHANNEL_SUPPORT: ch_reg = 2; + break; case FOUR_CHANNEL_SUPPORT: ch_reg = 1; + break; case TWO_CHANNEL_SUPPORT: ch_reg = 0; break; default: dev_err(dev->dev, "channel not supported\n"); + return -EINVAL; } i2s_disable_channels(dev, substream->stream);

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Re: [alsa-devel] [PATCH] Asoc: Samsung: Add support for I2S runtime S2R
by Mark Brown 26 Jan '13

26 Jan '13

On Thu, Jan 24, 2013 at 06:05:30PM +0530, R. Chandrasekar wrote: > From: "R. Chandrasekar" <rcsekar(a)samsung.com> > > This patch adds runtime suspend to resume support for I2S. > > Signed-off-by: Padmavathi Venna <padma.v(a)samsung.com> > Signed-off-by: R. Chandrasekar <rcsekar(a)samsung.com> This is good but doesn't apply against topic/samsung, please check and resend. Note that the signoffs should be in author order - the author should be first, then people forwarding on the patch. Other people should have other tags like Acked-by or Reviewed-by.

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Re: [alsa-devel] [PATCH] Asoc: Samsung: Add I2S S/W RST in startup function
by Mark Brown 26 Jan '13

26 Jan '13

On Thu, Jan 24, 2013 at 06:05:31PM +0530, R. Chandrasekar wrote: > From: Padmavathi Venna <padma.v(a)samsung.com> > > I2S module need to be reset after S2R. Keeping the S/W rst > control part in resume didn't help in playing audio after resume. > So this patch adds S/W RST control part in startup function which > gets triggered for every new audio stream playback. Applied, thanks. Please use subject lines matching the subsystem.

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Re: [alsa-devel] [PATCH 002/002] ASoC: Replace max98090 Device Driver
by Mark Brown 26 Jan '13

26 Jan '13

On Thu, Jan 24, 2013 at 03:28:54PM -0800, Jerry Wong wrote: > This is an odd control that is a combination of an enum and a power > management widget. The 0 value of the enum turns power off. > Not sure what is another choice. The enum value needs to be > saved so it can be restored when DAPM turns the control on. That doesn't appear to be what your control actually does - what happens if someone writes ot the control while the widget is powered down? > > +static const struct snd_kcontrol_new max98090_snd_controls[] = { > > + SOC_SINGLE("MIC Bias VCM Bandgap", M98090_REG_BIAS_CNTL, > > + M98090_VCM_MODE_SHIFT, M98090_VCM_MODE_NUM - 1, 0), > > Why is this a user visible control? > It is not controlled by DAPM. The low power function is taken care of by > /SHDN\, so this retains the user choice of bandgap or divider reference. Why would the user want this choice? > > +static int max98090_runtime_resume(struct device *dev) > > +{ > > + struct max98090_priv *max98090 = dev_get_drvdata(dev); > > + > > + regcache_cache_only(max98090->regmap, false); > > + > > + max98090_reset(max98090); > > + > > + regcache_sync(max98090->regmap); > > Do you really need to reset the device here? If anything I'd expect it > > on suspend since the reset state is probably the lowest power state. > Yes, for a soft reset where the device is not power cycled, DAPM has > the device go in and out of the low power /SHDN\ state, and the old > register values are retained. Without this, we would need to specify > that the PMIC power cycle the chip. This doesn't really answer my questions. Why would it matter if the PMIC has power cycled the chip, and if you do need to do this why is resume the time to do it?

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[alsa-devel] [PATCH 04/19] oss/dmabuf: use dma_map_single
by Arnd Bergmann 25 Jan '13

25 Jan '13

The virt_to_bus/bus_to_virt functions have been deprecated for as long as I can remember, and they are used in very few remaining instances, usually in obscure ISA device drivers. The OSS sound drivers are the only ones that are still used on the ARM architecture, and only on some of the earliest StrongARM machines. The problem for converting the OSS subsystem to use dma_map_single instead is that the caller of virt_to_bus does not have a device pointer, since the subsystem has never been ported to use the common device infrastructure. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: alsa-devel(a)alsa-project.org --- sound/oss/dmabuf.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/oss/dmabuf.c b/sound/oss/dmabuf.c index bcc3e8e..a59c888 100644 --- a/sound/oss/dmabuf.c +++ b/sound/oss/dmabuf.c @@ -114,7 +114,7 @@ static int sound_alloc_dmap(struct dma_buffparms *dmap) } } dmap->raw_buf = start_addr; - dmap->raw_buf_phys = virt_to_bus(start_addr); + dmap->raw_buf_phys = dma_map_single(NULL, start_addr, dmap->buffsize, DMA_BIDIRECTIONAL); for (page = virt_to_page(start_addr); page <= virt_to_page(end_addr); page++) SetPageReserved(page); @@ -139,6 +139,7 @@ static void sound_free_dmap(struct dma_buffparms *dmap) for (page = virt_to_page(start_addr); page <= virt_to_page(end_addr); page++) ClearPageReserved(page); + dma_unmap_single(NULL, dmap->raw_buf_phys, dmap->buffsize, DMA_BIDIRECTIONAL); free_pages((unsigned long) dmap->raw_buf, sz); dmap->raw_buf = NULL; } -- 1.8.0

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[alsa-devel] [PATCH] ASoC: wm_adsp: Accept 0 as a parameter block address
by Mark Brown 25 Jan '13

25 Jan '13

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 85 ++++++++++++++++++-------------------------- 1 file changed, 35 insertions(+), 50 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index bc5e383..3ba4237 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -589,27 +589,21 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) be32_to_cpu(adsp1_alg[i].dm), be32_to_cpu(adsp1_alg[i].zm)); - if (adsp1_alg[i].dm) { - region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; - region->type = WMFW_ADSP1_DM; - region->alg = be32_to_cpu(adsp1_alg[i].alg.id); - region->base = be32_to_cpu(adsp1_alg[i].dm); - list_add_tail(&region->list, - &dsp->alg_regions); - } + region = kzalloc(sizeof(*region), GFP_KERNEL); + if (!region) + return -ENOMEM; + region->type = WMFW_ADSP1_DM; + region->alg = be32_to_cpu(adsp1_alg[i].alg.id); + region->base = be32_to_cpu(adsp1_alg[i].dm); + list_add_tail(&region->list, &dsp->alg_regions); - if (adsp1_alg[i].zm) { - region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; - region->type = WMFW_ADSP1_ZM; - region->alg = be32_to_cpu(adsp1_alg[i].alg.id); - region->base = be32_to_cpu(adsp1_alg[i].zm); - list_add_tail(&region->list, - &dsp->alg_regions); - } + region = kzalloc(sizeof(*region), GFP_KERNEL); + if (!region) + return -ENOMEM; + region->type = WMFW_ADSP1_ZM; + region->alg = be32_to_cpu(adsp1_alg[i].alg.id); + region->base = be32_to_cpu(adsp1_alg[i].zm); + list_add_tail(&region->list, &dsp->alg_regions); break; case WMFW_ADSP2: @@ -623,38 +617,29 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) be32_to_cpu(adsp2_alg[i].ym), be32_to_cpu(adsp2_alg[i].zm)); - if (adsp2_alg[i].xm) { - region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; - region->type = WMFW_ADSP2_XM; - region->alg = be32_to_cpu(adsp2_alg[i].alg.id); - region->base = be32_to_cpu(adsp2_alg[i].xm); - list_add_tail(&region->list, - &dsp->alg_regions); - } + region = kzalloc(sizeof(*region), GFP_KERNEL); + if (!region) + return -ENOMEM; + region->type = WMFW_ADSP2_XM; + region->alg = be32_to_cpu(adsp2_alg[i].alg.id); + region->base = be32_to_cpu(adsp2_alg[i].xm); + list_add_tail(&region->list, &dsp->alg_regions); - if (adsp2_alg[i].ym) { - region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; - region->type = WMFW_ADSP2_YM; - region->alg = be32_to_cpu(adsp2_alg[i].alg.id); - region->base = be32_to_cpu(adsp2_alg[i].ym); - list_add_tail(&region->list, - &dsp->alg_regions); - } + region = kzalloc(sizeof(*region), GFP_KERNEL); + if (!region) + return -ENOMEM; + region->type = WMFW_ADSP2_YM; + region->alg = be32_to_cpu(adsp2_alg[i].alg.id); + region->base = be32_to_cpu(adsp2_alg[i].ym); + list_add_tail(&region->list, &dsp->alg_regions); - if (adsp2_alg[i].zm) { - region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; - region->type = WMFW_ADSP2_ZM; - region->alg = be32_to_cpu(adsp2_alg[i].alg.id); - region->base = be32_to_cpu(adsp2_alg[i].zm); - list_add_tail(&region->list, - &dsp->alg_regions); - } + region = kzalloc(sizeof(*region), GFP_KERNEL); + if (!region) + return -ENOMEM; + region->type = WMFW_ADSP2_ZM; + region->alg = be32_to_cpu(adsp2_alg[i].alg.id); + region->base = be32_to_cpu(adsp2_alg[i].zm); + list_add_tail(&region->list, &dsp->alg_regions); break; } } -- 1.7.10.4

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[alsa-devel] [PATCH 04/19] oss/dmabuf: use dma_map_single
by Arnd Bergmann 25 Jan '13

25 Jan '13

The virt_to_bus/bus_to_virt functions have been deprecated for as long as I can remember, and they are used in very few remaining instances, usually in obscure ISA device drivers. The OSS sound drivers are the only ones that are still used on the ARM architecture, and only on some of the earliest StrongARM machines. The problem for converting the OSS subsystem to use dma_map_single instead is that the caller of virt_to_bus does not have a device pointer, since the subsystem has never been ported to use the common device infrastructure. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: alsa-devel(a)alsa-project.org --- sound/oss/dmabuf.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/oss/dmabuf.c b/sound/oss/dmabuf.c index bcc3e8e..a59c888 100644 --- a/sound/oss/dmabuf.c +++ b/sound/oss/dmabuf.c @@ -114,7 +114,7 @@ static int sound_alloc_dmap(struct dma_buffparms *dmap) } } dmap->raw_buf = start_addr; - dmap->raw_buf_phys = virt_to_bus(start_addr); + dmap->raw_buf_phys = dma_map_single(NULL, start_addr, dmap->buffsize, DMA_BIDIRECTIONAL); for (page = virt_to_page(start_addr); page <= virt_to_page(end_addr); page++) SetPageReserved(page); @@ -139,6 +139,7 @@ static void sound_free_dmap(struct dma_buffparms *dmap) for (page = virt_to_page(start_addr); page <= virt_to_page(end_addr); page++) ClearPageReserved(page); + dma_unmap_single(NULL, dmap->raw_buf_phys, dmap->buffsize, DMA_BIDIRECTIONAL); free_pages((unsigned long) dmap->raw_buf, sz); dmap->raw_buf = NULL; } -- 1.8.0

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[alsa-devel] [PATCH] ASoC: arizona: Always keep SYSCLK link for DAIs
by Mark Brown 25 Jan '13

25 Jan '13

Don't disconnect SYSCLK when a DAI moves to the ASYNCCLK domain, we always need SYSCLK for operation even if it's not the reference clock for the domain. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 26 ++++++++++++++++++++------ 1 file changed, 20 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 27b6efc..f76e982 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -978,13 +978,27 @@ static int arizona_dai_set_sysclk(struct snd_soc_dai *dai, routes[0].sink = dai->driver->capture.stream_name; routes[1].sink = dai->driver->playback.stream_name; - routes[0].source = arizona_dai_clk_str(dai_priv->clk); - routes[1].source = arizona_dai_clk_str(dai_priv->clk); - snd_soc_dapm_del_routes(&codec->dapm, routes, ARRAY_SIZE(routes)); + switch (clk_id) { + case ARIZONA_CLK_SYSCLK: + routes[0].source = arizona_dai_clk_str(dai_priv->clk); + routes[1].source = arizona_dai_clk_str(dai_priv->clk); + snd_soc_dapm_del_routes(&codec->dapm, routes, + ARRAY_SIZE(routes)); + break; + default: + break; + } - routes[0].source = arizona_dai_clk_str(clk_id); - routes[1].source = arizona_dai_clk_str(clk_id); - snd_soc_dapm_add_routes(&codec->dapm, routes, ARRAY_SIZE(routes)); + switch (clk_id) { + case ARIZONA_CLK_ASYNCCLK: + routes[0].source = arizona_dai_clk_str(clk_id); + routes[1].source = arizona_dai_clk_str(clk_id); + snd_soc_dapm_add_routes(&codec->dapm, routes, + ARRAY_SIZE(routes)); + break; + default: + break; + } dai_priv->clk = clk_id; -- 1.7.10.4

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Re: [alsa-devel] [PATCH next-20130124] Sound: pci: Fix unused variable warning in patch_sigmatel.c
by Takashi Iwai 25 Jan '13

25 Jan '13

At Fri, 25 Jan 2013 00:38:06 +0200, Stratos Karafotis wrote: > > Fix the following build warnings > > sound/pci/hda/patch_sigmatel.c: In function ‘stac92hd71bxx_fixup_hp’: > sound/pci/hda/patch_sigmatel.c:2434:24: warning: unused variable ‘spec’ [-Wunused-variable] Thanks for catching. But such temporal cast like your patch makes the code readability much worse. The code readability (thus its maintainability) is a more important factor. I'm going to fix the definition of snd_printd() itself for fixing this kind of warnings. We can use inline functions instead of empty macros to achieve the same but without triggering compiler warnings. Takashi > > Signed-off-by: Stratos Karafotis <stratosk(a)semaphore.gr> > --- > sound/pci/hda/patch_sigmatel.c | 14 +++++--------- > 1 file changed, 5 insertions(+), 9 deletions(-) > > diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c > index 0aa0ceb..f269d1f 100644 > --- a/sound/pci/hda/patch_sigmatel.c > +++ b/sound/pci/hda/patch_sigmatel.c > @@ -2365,7 +2365,6 @@ static void stac92hd71bxx_fixup_ref(struct hda_codec *codec, > static void stac92hd71bxx_fixup_hp_m4(struct hda_codec *codec, > const struct hda_fixup *fix, int action) > { > - struct sigmatel_spec *spec = codec->spec; > struct hda_jack_tbl *jack; > > if (action != HDA_FIXUP_ACT_PRE_PROBE) > @@ -2381,7 +2380,7 @@ static void stac92hd71bxx_fixup_hp_m4(struct hda_codec *codec, > if (jack) > jack->private_data = 0x02; > > - spec->gpio_mask |= 0x02; > + ((struct sigmatel_spec *)codec->spec)->gpio_mask |= 0x02; > > /* enable internal microphone */ > snd_hda_codec_set_pincfg(codec, 0x0e, 0x01813040); > @@ -2420,19 +2419,15 @@ static void stac92hd71bxx_fixup_hp_dv5(struct hda_codec *codec, > static void stac92hd71bxx_fixup_hp_hdx(struct hda_codec *codec, > const struct hda_fixup *fix, int action) > { > - struct sigmatel_spec *spec = codec->spec; > - > if (action != HDA_FIXUP_ACT_PRE_PROBE) > return; > - spec->gpio_led = 0x08; > + ((struct sigmatel_spec *)codec->spec)->gpio_led = 0x08; > } > > > static void stac92hd71bxx_fixup_hp(struct hda_codec *codec, > const struct hda_fixup *fix, int action) > { > - struct sigmatel_spec *spec = codec->spec; > - > if (action != HDA_FIXUP_ACT_PRE_PROBE) > return; > > @@ -2456,8 +2451,9 @@ static void stac92hd71bxx_fixup_hp(struct hda_codec *codec, > > if (find_mute_led_cfg(codec, 1)) > snd_printd("mute LED gpio %d polarity %d\n", > - spec->gpio_led, > - spec->gpio_led_polarity); > + ((struct sigmatel_spec *)codec->spec)->gpio_led, > + ((struct sigmatel_spec *)codec->spec)-> > + gpio_led_polarity); > > } > > -- > 1.8.1 >

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