December 2011 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] [PATCH] ASoC: WM8958: correctly show firmware magic on mismatch
by Mark Brown 12 Dec '11

12 Dec '11

On Sun, Dec 11, 2011 at 07:15:49PM +0100, Jonathan Neuschäfer wrote: > Signed-off-by: Jonathan Neuschäfer <j.neuschaefer(a)gmx.net> Applied, thanks.

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Re: [alsa-devel] [PATCH 07/10] ASoC: OMAP: ams-delta: Drive modem/codec pins over GPIO API
by Mark Brown 12 Dec '11

12 Dec '11

On Sun, Dec 11, 2011 at 09:12:05PM +0100, Janusz Krzysztofik wrote: > Don't use Amstrad Delta custom I/O functions any longer, replace them > with GPIO. Old pin definitions, no longer used by the modem bits either, > can be dropped. Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com>

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[alsa-devel] [PATCH] ASoC: Raise Speyside audio system clock rate to 512fs
by Mark Brown 11 Dec '11

11 Dec '11

To support advanced system functionality for additional components; the actively used clocks will remain the same for current components. Also factor the rate out to a single #define while we're at it. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/speyside.c | 7 ++++--- 1 files changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index e0f55cb5..8cdd4f4 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -19,6 +19,7 @@ #include "../codecs/wm9081.h" #define WM8996_HPSEL_GPIO 214 +#define MCLK_AUDIO_RATE (512 * 48000) static int speyside_set_bias_level(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, @@ -67,7 +68,7 @@ static int speyside_set_bias_level_post(struct snd_soc_card *card, if (card->dapm.bias_level == SND_SOC_BIAS_STANDBY) { ret = snd_soc_dai_set_pll(codec_dai, 0, WM8996_FLL_MCLK2, - 32768, 48000 * 256); + 32768, MCLK_AUDIO_RATE); if (ret < 0) { pr_err("Failed to start FLL\n"); return ret; @@ -75,7 +76,7 @@ static int speyside_set_bias_level_post(struct snd_soc_card *card, ret = snd_soc_dai_set_sysclk(codec_dai, WM8996_SYSCLK_FLL, - 48000 * 256, + MCLK_AUDIO_RATE, SND_SOC_CLOCK_IN); if (ret < 0) return ret; @@ -197,7 +198,7 @@ static int speyside_wm9081_init(struct snd_soc_dapm_context *dapm) { /* At any time the WM9081 is active it will have this clock */ return snd_soc_codec_set_sysclk(dapm->codec, WM9081_SYSCLK_MCLK, 0, - 48000 * 256, 0); + MCLK_AUDIO_RATE, 0); } static struct snd_soc_aux_dev speyside_aux_dev[] = { -- 1.7.7.3

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[alsa-devel] [PATCH 0/5] ASoC: OMAP4: Device tree support for DMIC, McPDM
by Peter Ujfalusi 11 Dec '11

11 Dec '11

Hello, The following series adds device tree support for omap-dmic, omap-mcpdm ASoC DAI drivers. The OMAP4 platform patches have dependency on the DMIC driver series: http://www.mail-archive.com/linux-omap@vger.kernel.org/msg59161.html Regards, Peter --- Peter Ujfalusi (5): ASoC: omap-dmic: Add device tree bindings ASoC: omap-mcpdm: Add device tree bindings OMAP4: devices: Do not create dmic device if the dtb has been provided OMAP4: devices: Do not create mcpdm device if the dtb has been provided ARM: OMAP4: DTS: Support for dmic, and McPDM in device tree .../bindings/sound/soc/omap/omap-dmic.txt | 13 +++++++++++++ .../bindings/sound/soc/omap/omap-mcpdm.txt | 13 +++++++++++++ arch/arm/boot/dts/omap4-panda.dts | 4 ++++ arch/arm/boot/dts/omap4.dtsi | 9 +++++++++ arch/arm/mach-omap2/devices.c | 8 ++++++++ sound/soc/omap/omap-dmic.c | 8 ++++++++ sound/soc/omap/omap-mcpdm.c | 8 ++++++++ 7 files changed, 63 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/soc/omap/omap-dmic.txt create mode 100644 Documentation/devicetree/bindings/sound/soc/omap/omap-mcpdm.txt -- 1.7.8.rc4

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Re: [alsa-devel] [PATCH] ASoC: Fix an obvious copy paste error in an error message
by Mark Brown 11 Dec '11

11 Dec '11

On Fri, Dec 09, 2011 at 02:16:29PM +0100, Lothar Waßmann wrote: > The message was obviously copied from soc_init_codec_debugfs() Applied, thanks.

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[alsa-devel] [PATCH] ASoC: Fix WM8996 24.576MHz clock operation
by Mark Brown 10 Dec '11

10 Dec '11

Record the clock after the divider as that is what all SYSCLK users see. Without this the other clock configuration in the device comes out at half rate. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 41cc9d2..8f88f5a 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2043,6 +2043,7 @@ static int wm8996_set_sysclk(struct snd_soc_dai *dai, break; case 24576000: ratediv = WM8996_SYSCLK_DIV; + wm8996->sysclk /= 2; case 12288000: snd_soc_update_bits(codec, WM8996_AIF_RATE, WM8996_SYSCLK_RATE, WM8996_SYSCLK_RATE); -- 1.7.7.3

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Re: [alsa-devel] [PATCH 0/5] RFC v2 for snd-usb endpoint logic rework
by Aurélien Leblond 09 Dec '11

09 Dec '11

Hi Daniel, Compiled and running! So far so good! I haven't pushed the jack parameters more than I would usually though... I'm still at 16ms latency (Frame/period: 256 ; Sample Rate: 48000 ; Periods Buffer: 3). That was the minimum I could go with the previous driver, so I'll try to push it a bit tomorrow and see how it goes. Is there any particular testing you would like to see done? Another thing, in your previous post, you asked to try the driver with other sound card. I have an Edirol UA-25Ex that never worked on this machine (I have the dredded EHCI USB port, it's the reason why I bought the FTU in the first place). The error I used to get while starting Jack in Full Duplex with this sound card was "Cannot submit datapipe for urb 0, error -28: not enough bandwidth" But, "apparently".......now it works! I tried it not later than last week with Kernel 3.0 and it still wasn't working. But with 3.2 and your patch, (and after running the FTU on the very same USB port), I got the UA-25Ex to work in full duplex for the first time ever. I don't know if it is due to the new version of the kernel, or your patch, or something else, but this is huge because a lot of people used to have this problem with EHCI and USB 1 sound cards. (or may be it was simply fixed between 3.0 and 3.2 and I simply missed it!). So the second question is: what do you think? Any info you would like to see posted here? (dmesg trace....anything else?). Anyway, thanks a lot for everything :) Aurélien On Thu, Dec 8, 2011 at 7:38 PM, Daniel Mack <zonque(a)gmail.com> wrote: > Hi Aurélien, > > On 12/08/2011 08:33 PM, Aurélien Leblond wrote: >> I finally got some time to test this driver! >> I'm not sure I understand where to get Takashi's sound tree. >> >> I tried to apply your patch against a clone of >> git://git.alsa-project.org/alsa-kmirror.git alsa-kmirror, but then >> GitCompile failed. >> >> I then tried to use >> git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git , but >> I'm not even sure what I'm pulling to be honest... >> >> Would you mind just guiding me on how to prepare the source code? > > No problem :) > > Clone Linus' tree: > > git://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git > > Make a local branch and pull in Takashi's tree: > > cd linux > git checkout -b alsa > git pull git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git > master > > Make another branch on top of that > > git checkout -b usbaudio > > Then apply my patch(es). For the next round, just kill the local > usbaudio branch again and start over: > > git reset --hard > git checkout alsa > git branch -D usbaudio > git checkout -b usbaudio > > Then apply the next round :) > >> Thanks in advance, and thanks for all the work you have already done! > > No problem. I'm planning to continue work on this these days. It's just > that I'm constantly distracted by other things :-/ > > Daniel

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[alsa-devel] [PATCH v4] ASoC: Allow DAI links to be specified using device tree nodes
by Stephen Warren 09 Dec '11

09 Dec '11

DAI link endpoints and platform (DMA) devices are currently specified by name. When instantiating sound cards from device tree, it may be more convenient to refer to these devices by phandle in the device tree, and for code to describe DAI links using the "struct device_node *" ("of_node") those phandles map to. This change adds new fields to snd_soc_dai_link which can "name" devices using of_node, enhances soc_bind_dai_link() to allow binding based on of_node, and enhances snd_soc_register_card() to ensure that illegal combinations of name and of_node are not used. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> Acked-by: Liam Girdwood <lrg(a)ti.com> COMMIT NOTES: v2: New patch implementing a new feature v3: Add Liam's ack v4: Rebased to re-order this patch directly onto ASoC for-next --- include/sound/soc.h | 4 +++ sound/soc/soc-core.c | 64 ++++++++++++++++++++++++++++++++++++++++++++----- 2 files changed, 61 insertions(+), 7 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index d9aa66b..40c256e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -231,6 +231,7 @@ enum snd_soc_bias_level { SND_SOC_BIAS_ON = 3, }; +struct device_node; struct snd_jack; struct snd_soc_card; struct snd_soc_pcm_stream; @@ -704,8 +705,11 @@ struct snd_soc_dai_link { const char *name; /* Codec name */ const char *stream_name; /* Stream name */ const char *codec_name; /* for multi-codec */ + const struct device_node *codec_of_node; const char *platform_name; /* for multi-platform */ + const struct device_node *platform_of_node; const char *cpu_dai_name; + const struct device_node *cpu_dai_of_node; const char *codec_dai_name; unsigned int dai_fmt; /* format to set on init */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ebb1048..56792d4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -763,8 +763,13 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) } /* no, then find CPU DAI from registered DAIs*/ list_for_each_entry(cpu_dai, &dai_list, list) { - if (strcmp(cpu_dai->name, dai_link->cpu_dai_name)) - continue; + if (dai_link->cpu_dai_of_node) { + if (cpu_dai->dev->of_node != dai_link->cpu_dai_of_node) + continue; + } else { + if (strcmp(cpu_dai->name, dai_link->cpu_dai_name)) + continue; + } rtd->cpu_dai = cpu_dai; goto find_codec; @@ -780,8 +785,13 @@ find_codec: /* no, then find CODEC from registered CODECs*/ list_for_each_entry(codec, &codec_list, list) { - if (strcmp(codec->name, dai_link->codec_name)) - continue; + if (dai_link->codec_of_node) { + if (codec->dev->of_node != dai_link->codec_of_node) + continue; + } else { + if (strcmp(codec->name, dai_link->codec_name)) + continue; + } rtd->codec = codec; @@ -813,13 +823,19 @@ find_platform: /* if there's no platform we match on the empty platform */ platform_name = dai_link->platform_name; - if (!platform_name) + if (!platform_name && !dai_link->platform_of_node) platform_name = "snd-soc-dummy"; /* no, then find one from the set of registered platforms */ list_for_each_entry(platform, &platform_list, list) { - if (strcmp(platform->name, platform_name)) - continue; + if (dai_link->platform_of_node) { + if (platform->dev->of_node != + dai_link->platform_of_node) + continue; + } else { + if (strcmp(platform->name, platform_name)) + continue; + } rtd->platform = platform; goto out; @@ -2830,6 +2846,40 @@ int snd_soc_register_card(struct snd_soc_card *card) if (!card->name || !card->dev) return -EINVAL; + for (i = 0; i < card->num_links; i++) { + struct snd_soc_dai_link *link = &card->dai_link[i]; + + /* + * Codec must be specified by 1 of name or OF node, + * not both or neither. + */ + if (!!link->codec_name == !!link->codec_of_node) { + dev_err(card->dev, + "Neither/both codec name/of_node are set\n"); + return -EINVAL; + } + + /* + * Platform may be specified by either name or OF node, but + * can be left unspecified, and a dummy platform will be used. + */ + if (link->platform_name && link->platform_of_node) { + dev_err(card->dev, + "Both platform name/of_node are set\n"); + return -EINVAL; + } + + /* + * CPU DAI must be specified by 1 of name or OF node, + * not both or neither. + */ + if (!!link->cpu_dai_name == !!link->cpu_dai_of_node) { + dev_err(card->dev, + "Neither/both cpu_dai name/of_node are set\n"); + return -EINVAL; + } + } + dev_set_drvdata(card->dev, card); snd_soc_initialize_card_lists(card); -- 1.7.0.4

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[alsa-devel] [PATCH] ASoC: Fix comments for disabling amplifier and PGA
by Axel Lin 09 Dec '11

09 Dec '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8961.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index dc087c1..58fbf0a 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -422,11 +422,11 @@ static int wm8961_spk_event(struct snd_soc_dapm_widget *w, } if (event & SND_SOC_DAPM_PRE_PMD) { - /* Enable the amplifier */ + /* Disable the amplifier */ spk_reg &= ~(WM8961_SPKL_ENA | WM8961_SPKR_ENA); snd_soc_write(codec, WM8961_CLASS_D_CONTROL_1, spk_reg); - /* Enable the PGA */ + /* Disable the PGA */ pwr_reg &= ~(WM8961_SPKL_PGA | WM8961_SPKR_PGA); snd_soc_write(codec, WM8961_PWR_MGMT_2, pwr_reg); } -- 1.7.5.4

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[alsa-devel] [PATCH 1/4] ASoC: Convert WM8776 to devm_kzalloc()
by Mark Brown 09 Dec '11

09 Dec '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8776.c | 14 ++++++-------- 1 files changed, 6 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index fbf80c5..38b4556 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -452,7 +452,8 @@ static int __devinit wm8776_spi_probe(struct spi_device *spi) struct wm8776_priv *wm8776; int ret; - wm8776 = kzalloc(sizeof(struct wm8776_priv), GFP_KERNEL); + wm8776 = devm_kzalloc(&spi->dev, sizeof(struct wm8776_priv), + GFP_KERNEL); if (wm8776 == NULL) return -ENOMEM; @@ -461,15 +462,13 @@ static int __devinit wm8776_spi_probe(struct spi_device *spi) ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm8776, wm8776_dai, ARRAY_SIZE(wm8776_dai)); - if (ret < 0) - kfree(wm8776); + return ret; } static int __devexit wm8776_spi_remove(struct spi_device *spi) { snd_soc_unregister_codec(&spi->dev); - kfree(spi_get_drvdata(spi)); return 0; } @@ -491,7 +490,8 @@ static __devinit int wm8776_i2c_probe(struct i2c_client *i2c, struct wm8776_priv *wm8776; int ret; - wm8776 = kzalloc(sizeof(struct wm8776_priv), GFP_KERNEL); + wm8776 = devm_kzalloc(&i2c->dev, sizeof(struct wm8776_priv), + GFP_KERNEL); if (wm8776 == NULL) return -ENOMEM; @@ -500,15 +500,13 @@ static __devinit int wm8776_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8776, wm8776_dai, ARRAY_SIZE(wm8776_dai)); - if (ret < 0) - kfree(wm8776); + return ret; } static __devexit int wm8776_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); return 0; } -- 1.7.7.3

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