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

[alsa-devel] [PATCH] ASoC: Change spitz_ext_control to take dapm as argument.
by Axel Lin 17 Feb '12

17 Feb '12

This fixes below build warning: CC sound/soc/pxa/spitz.o sound/soc/pxa/spitz.c: In function 'spitz_startup': sound/soc/pxa/spitz.c:116: warning: passing argument 1 of 'spitz_ext_control' from incompatible pointer type sound/soc/pxa/spitz.c:47: note: expected 'struct snd_soc_card *' but argument is of type 'struct snd_soc_codec *' Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/pxa/spitz.c | 12 +++++------- 1 files changed, 5 insertions(+), 7 deletions(-) diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 4d108dc..fc052d8 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -44,10 +44,8 @@ static int spitz_jack_func; static int spitz_spk_func; static int spitz_mic_gpio; -static void spitz_ext_control(struct snd_soc_card *card) +static void spitz_ext_control(struct snd_soc_dapm_context *dapm) { - struct snd_soc_dapm_context *dapm = &card->dapm; - if (spitz_spk_func == SPITZ_SPK_ON) snd_soc_dapm_enable_pin(dapm, "Ext Spk"); else @@ -113,7 +111,7 @@ static int spitz_startup(struct snd_pcm_substream *substream) mutex_lock(&codec->mutex); /* check the jack status at stream startup */ - spitz_ext_control(codec); + spitz_ext_control(&codec->dapm); mutex_unlock(&codec->mutex); @@ -179,7 +177,7 @@ static int spitz_set_jack(struct snd_kcontrol *kcontrol, return 0; spitz_jack_func = ucontrol->value.integer.value[0]; - spitz_ext_control(card); + spitz_ext_control(&card->dapm); return 1; } @@ -193,13 +191,13 @@ static int spitz_get_spk(struct snd_kcontrol *kcontrol, static int spitz_set_spk(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); if (spitz_spk_func == ucontrol->value.integer.value[0]) return 0; spitz_spk_func = ucontrol->value.integer.value[0]; - spitz_ext_control(card); + spitz_ext_control(&card->dapm); return 1; } -- 1.7.5.4

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[alsa-devel] [PATCH 1/2] ASoC: core - Add direct mapping between DAIs and DAPM
by Liam Girdwood 17 Feb '12

17 Feb '12

Currently we map DAI streams to DAPM widgets based on a string stream name. This can lead to some limitations when working with lots of DAI links as there is a risk of naming collision with similar stream names. This patch creates a 1:N mapping between a DAI and it's DAPM widgets. It's still string based, but we now directly reference the widget names with the DAI and bind the DAI to the widget ptrs at initialisation. The binding also tales into consideration the parent codec or platform that the DAI and widget belong to so that we can easily support multiple components with the same widget names. This patch still works in conjunction with the stream based events, but it is intended to eventually replace the DAI stream name. e.g. to map twl6040 headset DACs to headset DAI static const char *headset_widgets[] = {"HSDAC Left", "HSDAC Right"}; static struct snd_soc_dai_driver twl6040_dai[] = { ... { .name = "twl6040-dl1", .playback = { .stream_name = "Headset Playback", .channel_widgets = headset_widgets, .channels_min = 1, .channels_max = ARRAY_SIZE(headset_widgets), .rates = TWL6040_RATES, .formats = TWL6040_FORMATS, }, .ops = &twl6040_dai_ops, }, This patch will allow DAI connection to any widget type as not all DAIs connect directly to DACs or ADCs, some can connect via a mux that can swap channels etc. This patch is also intended to be the starting point for future DAPM channel mapping support as we now have a direct link between PCMs and DAPM widgets. e.g. In the example above we map channel 0 (left) to "HSDAC Left" and channel 1 (right) to "HSDAC Right". Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc-dai.h | 11 ++- include/sound/soc.h | 1 + sound/soc/soc-core.c | 42 +++++++++++++ sound/soc/soc-dapm.c | 155 +++++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 205 insertions(+), 4 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 2413acc..ce5beea 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -246,14 +246,17 @@ struct snd_soc_dai { unsigned int rate; /* parent platform/codec */ - union { - struct snd_soc_platform *platform; - struct snd_soc_codec *codec; - }; + struct snd_soc_platform *platform; + struct snd_soc_codec *codec; struct snd_soc_card *card; struct list_head list; struct list_head card_list; + + /* runtime widgets */ + struct snd_soc_dapm_widget **playback_widgets; + struct snd_soc_dapm_widget **capture_widgets; + bool dapm_bind_complete; }; static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai, diff --git a/include/sound/soc.h b/include/sound/soc.h index 8fa4dca..bace9b8 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -507,6 +507,7 @@ struct snd_soc_pcm_stream { unsigned int rates; /* SNDRV_PCM_RATE_* */ unsigned int rate_min; /* min rate */ unsigned int rate_max; /* max rate */ + const char **channel_widgets; /* DAPM channel widget names */ unsigned int channels_min; /* min channels */ unsigned int channels_max; /* max channels */ unsigned int sig_bits; /* number of bits of content */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6bad7cd..733f5de 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -68,6 +68,8 @@ static int pmdown_time = 5000; module_param(pmdown_time, int, 0); MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)"); +int dapm_bind_dai_widgets(struct snd_soc_dai *dai, int stream); + /* returns the minimum number of bytes needed to represent * a particular given value */ static int min_bytes_needed(unsigned long val) @@ -1392,6 +1394,41 @@ static int snd_soc_init_codec_cache(struct snd_soc_codec *codec, return 0; } +static int soc_init_card_dai_widgets(struct snd_soc_card *card) +{ + struct snd_soc_dai *dai; + int ret = 0; + + list_for_each_entry(dai, &card->dai_dev_list, card_list) { + + /* only process DAIs that use the new API until + * the old "stream name" API is fully deprecated */ + if (!dai->driver->playback.channel_widgets && + !dai->driver->capture.channel_widgets) + continue; + + /* skip if already instanciated */ + if (dai->dapm_bind_complete) + continue; + + /* bind DAI widget name to component widget */ + if (dai->driver->playback.channels_max) { + ret = dapm_bind_dai_widgets(dai, SNDRV_PCM_STREAM_PLAYBACK); + if (ret < 0) + return ret; + } + if (dai->driver->capture.channels_max) { + ret = dapm_bind_dai_widgets(dai, SNDRV_PCM_STREAM_CAPTURE); + if (ret < 0) + return ret; + } + + dai->dapm_bind_complete = 1; + } + + return ret; +} + static void snd_soc_instantiate_card(struct snd_soc_card *card) { struct snd_soc_codec *codec; @@ -1556,6 +1593,9 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) } snd_soc_dapm_new_widgets(&card->dapm); + ret = soc_init_card_dai_widgets(card); + if (ret < 0) + goto probe_aux_dev_err; if (card->fully_routed) list_for_each_entry(codec, &card->codec_dev_list, card_list) @@ -3139,6 +3179,8 @@ found: pr_debug("Unregistered DAI '%s'\n", dai->name); kfree(dai->name); + kfree(dai->playback_widgets); + kfree(dai->capture_widgets); kfree(dai); } EXPORT_SYMBOL_GPL(snd_soc_unregister_dai); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 63a5614..5f99842 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -829,6 +829,161 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) return con; } +static struct snd_soc_dapm_widget * + dapm_get_codec_widget(struct snd_soc_codec *codec, const char *name) +{ + struct snd_soc_card *card = codec->card; + struct snd_soc_dapm_widget *w; + + /* find widget for this codec and widget name */ + list_for_each_entry(w, &card->widgets, list) { + + /* make sure the widget belongs to the DAI codec */ + if (w->codec && w->codec != codec) + continue; + + if (!strcmp(w->name, name)) + return w; + } + + return NULL; +} + +static struct snd_soc_dapm_widget * + dapm_get_platform_widget(struct snd_soc_platform *platform, + const char *name) +{ + struct snd_soc_card *card = platform->card; + struct snd_soc_dapm_widget *w; + + /* find widget for this platform and widget name */ + list_for_each_entry(w, &card->widgets, list) { + + /* make sure the widget belongs to the DAI platform */ + if (w->platform && w->platform != platform) + continue; + + if (!strcmp(w->name, name)) + return w; + } + + return NULL; +} + +int dapm_bind_dai_playback_widgets(struct snd_soc_dai *dai) +{ + struct snd_soc_card *card = dai->card; + struct snd_soc_dapm_widget *w = NULL; + const char *widget_name; + int i; + + if (!dai->driver->playback.channel_widgets) + return 0; + + dai->playback_widgets = + kzalloc(sizeof(struct snd_soc_dapm_widget *) * + dai->driver->playback.channels_max, GFP_KERNEL); + if (!dai->playback_widgets) + return -ENOMEM; + + for (i = 0; i < dai->driver->playback.channels_max; i++) { + widget_name = dai->driver->playback.channel_widgets[i]; + + if (dai->codec) { + struct snd_soc_codec *codec; + + list_for_each_entry(codec, &card->codec_dev_list, card_list) { + w = dapm_get_codec_widget(codec, widget_name); + if (w) { + dev_dbg(dai->dev, "bind %s to %s\n", w->name, codec->name); + break; + } + } + } else if (dai->platform) { + struct snd_soc_platform *platform; + + list_for_each_entry(platform, &card->platform_dev_list, card_list) { + w = dapm_get_platform_widget(platform, widget_name); + if (w) { + dev_dbg(dai->dev, "bind %s to %s\n", w->name, platform->name); + break; + } + } + } + + if (w) + dai->playback_widgets[i] = w; + else { + dev_err(dai->dev, "unable to find playback DAI AIF %s\n", + widget_name); + return -EINVAL; + } + } + + return 0; +} + +int dapm_bind_dai_capture_widgets(struct snd_soc_dai *dai) +{ + struct snd_soc_card *card = dai->card; + struct snd_soc_dapm_widget *w = NULL; + const char *widget_name; + int i; + + if (!dai->driver->capture.channel_widgets) + return 0; + + dai->capture_widgets = + kzalloc(sizeof(struct snd_soc_dapm_widget *) * + dai->driver->capture.channels_max, GFP_KERNEL); + if (!dai->capture_widgets) + return -ENOMEM; + + for (i = 0; i < dai->driver->capture.channels_max; i++) { + widget_name = dai->driver->capture.channel_widgets[i]; + + if (dai->codec) { + struct snd_soc_codec *codec; + + list_for_each_entry(codec, &card->codec_dev_list, card_list) { + w = dapm_get_codec_widget(codec, widget_name); + if (w) { + dev_dbg(dai->dev, "bind %s to %s\n", w->name, codec->name); + break; + } + } + } else if (dai->platform) { + struct snd_soc_platform *platform; + + list_for_each_entry(platform, &card->platform_dev_list, card_list) { + w = dapm_get_platform_widget(platform, widget_name); + if (w) { + dev_dbg(dai->dev, "bind %s to %s\n", w->name, platform->name); + break; + } + } + } + + if (w) + dai->capture_widgets[i] = w; + else { + dev_err(dai->dev, "unable to find playback DAI AIF %s\n", + widget_name); + return -EINVAL; + } + } + + return 0; +} + +int dapm_bind_dai_widgets(struct snd_soc_dai *dai, int stream) +{ + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + return dapm_bind_dai_playback_widgets(dai); + else + return dapm_bind_dai_capture_widgets(dai); +} + /* * Handler for generic register modifier widget. */ -- 1.7.5.4

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[alsa-devel] Realtime controlling of ALSA parameters from UI
by Aleksey Midenkov 17 Feb '12

17 Feb '12

Hello guys! I want to suggest you an idea of what I am thinking for a long time. And want to know of what you are thinking on it from perspective of how hard it would be to implement. I'm pretty sure the feature will be well popular because for me as an ordinary user it would be great stuff! So to the point. What I am saying about is the ability to control parameters from asound.conf at user level. This would give us such great things like switching sound output realtime or control each channel volume level in mixer GUI. If you will suggest userpace daemons like Pulse or Jack. There are some situations (like playing piano f.ex.) where realtime really matters, where such daemons put unacceptable latency. Another point is that they are not such feature rich. They cannot downmix f.ex. with controlling of each channel volume level (please tell me if I am wrong here). Every opinion is appreciated!

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[alsa-devel] [PATCH] ALSA: core: Constify the name in new kcontrols
by Mark Brown 17 Feb '12

17 Feb '12

We never modify it and this lets us use a const string as the name without warnings. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- I've got some ASoC patches which need this to avoid warnings, if you're OK with it can I apply it via there please? include/sound/control.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/include/sound/control.h b/include/sound/control.h index b2796e8..57815f6 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -40,7 +40,7 @@ struct snd_kcontrol_new { snd_ctl_elem_iface_t iface; /* interface identifier */ unsigned int device; /* device/client number */ unsigned int subdevice; /* subdevice (substream) number */ - unsigned char *name; /* ASCII name of item */ + const unsigned char *name; /* ASCII name of item */ unsigned int index; /* index of item */ unsigned int access; /* access rights */ unsigned int count; /* count of same elements */ -- 1.7.9.rc1

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[alsa-devel] Usage of jack_status_check in ASoC GPIO
by Omair Mohammed Abdullah 17 Feb '12

17 Feb '12

If I need to call a codec driver function whenever Headset status changes (to set various registers, turn off micbias etc.). In such a case, can the jack_status_check callback in snd_soc_jack_gpio structure be used? If I use it, I would need to modify the callback to pass certain parameters (codec pointer, status). Is this the intended use of the callback?

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[alsa-devel] [PATCH 1/4] ASoC: ak4535: Remove -codec from driver name
by Mark Brown 17 Feb '12

17 Feb '12

Redundant, the device is only a CODEC. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/ak4535.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index dd15516..29d4dec 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -443,7 +443,7 @@ MODULE_DEVICE_TABLE(i2c, ak4535_i2c_id); static struct i2c_driver ak4535_i2c_driver = { .driver = { - .name = "ak4535-codec", + .name = "ak4535", .owner = THIS_MODULE, }, .probe = ak4535_i2c_probe, -- 1.7.9.rc1

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[alsa-devel] [PATCH 1/3] ASoC: wm8962: Convert to module_i2c_driver()
by Mark Brown 17 Feb '12

17 Feb '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 12 +----------- 1 files changed, 1 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 2f06886..bc04b49 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3793,17 +3793,7 @@ static struct i2c_driver wm8962_i2c_driver = { .id_table = wm8962_i2c_id, }; -static int __init wm8962_modinit(void) -{ - return i2c_add_driver(&wm8962_i2c_driver); -} -module_init(wm8962_modinit); - -static void __exit wm8962_exit(void) -{ - i2c_del_driver(&wm8962_i2c_driver); -} -module_exit(wm8962_exit); +module_i2c_driver(wm8962_i2c_driver); MODULE_DESCRIPTION("ASoC WM8962 driver"); MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); -- 1.7.9.rc1

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[alsa-devel] [PATCH 1/7] ASoC: wm8962: Update the clocking when setting system clock
by Mark Brown 17 Feb '12

17 Feb '12

Make sure we update for any changes in cases where we reconfigure while live (eg, for analogue bypass). Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 4a081dd..3f37a91 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2705,6 +2705,8 @@ static int wm8962_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, wm8962->sysclk_rate = freq; + wm8962_configure_bclk(codec); + return 0; } -- 1.7.9.rc1

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[alsa-devel] snd_emu10k1 with E-MU 1616m PCIe
by tilt 16 Feb '12

16 Feb '12

Hello! I use the Creative E-MU 1616m PCIe: 03:04.0 Multimedia audio controller: Creative Labs SB0400 Audigy2 Value Subsystem: Creative Labs Device 4007 I am on Linux kernel 3.2.4, -driver and -firmware from git; -utils from 1.0.24.2-5 (Debian amd_64). I have manually applied the patch from [1] and initialized the device successfully: [ 3.387211] snd_emu10k1 0000:03:04.0: PCI INT A -> GSI 16 (level, low) -> IRQ 16 [ 3.387341] ALSA emu10k1_main.c:823 emu1010: Special config. [ 3.387479] ALSA emu10k1_main.c:862 emu1010: EMU_HANA_ID = 0x3f [ 3.387530] ALSA emu10k1_main.c:881 emu1010: filename emu/emu1010b.fw testing [ 3.393206] ALSA emu10k1_main.c:686 firmware size = 0x11bdc [ 6.294376] ALSA emu10k1_main.c:898 emu1010: Hana Firmware loaded [ 6.294475] ALSA emu10k1_main.c:901 emu1010: Hana version: 1.6 [ 6.294583] ALSA emu10k1_main.c:906 emu1010: Card options = 0x8 [ 6.294656] ALSA emu10k1_main.c:908 emu1010: Card options = 0x8 [ 6.295222] ALSA emu10k1_main.c:947 emu1010: Card options3 = 0x8 [ 6.314806] ALSA emu10k1_main.c:227 Audigy2 value: Special config. [ 6.315231] ALSA emufx.c:1545 EMU outputs on [ 6.315280] ALSA emufx.c:1593 EMU2 inputs on There are irregularities & limitations, but I do have 48kHz capture and playback. I have two questions: 1.: I only use Dock ports, and I notice many irregularities in DAC and ADC assignment; so I wonder: should /usr/local/lib/firmware/emu/micro_dock.fw not also be loaded? 2.: I use JACK with input: hw:0,2 (Mutichannel Capture/PT Playback) output: hw:0,3 (Multichannel Playback) One thing that I would like to fix if at all possible: I can not assign anything to Dock Phones L/R; instead they always repeat Dock DAC 1L/R, no matter what I route there. Is this a limitation of the firmware, or are the controls misleading me? Kind regards & thanks in advance for any suggestions, Tilman. [1] ALSA bugtracking system 0005250: E-MU 1010 PCI-Express not recognized https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5250

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[alsa-devel] [PATCH] ASoC: dapm: Only mark pin widgets as dirty if we actually change state
by Mark Brown 16 Feb '12

16 Feb '12

Small optimisation for noop state updates. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 2844879..a2a9c11 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1915,10 +1915,12 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, return -EINVAL; } + if (w->connected != status) + dapm_mark_dirty(w, "pin configuration"); + w->connected = status; if (status == 0) w->force = 0; - dapm_mark_dirty(w, "pin configuration"); return 0; } -- 1.7.9.rc1

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