September 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: Fix incorrect register cache size configuration
by Dimitris Papastamos 11 Sep '10

11 Sep '10

The reg_cache_size is the number of elements in the register cache, not the size of the cache itself. This is not a problem if the size of each element of the cache is 1 byte but it matters in any other case. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8580.c | 2 +- sound/soc/codecs/wm8711.c | 2 +- sound/soc/codecs/wm8728.c | 2 +- sound/soc/codecs/wm8731.c | 2 +- sound/soc/codecs/wm8741.c | 2 +- sound/soc/codecs/wm8750.c | 2 +- sound/soc/codecs/wm8753.c | 2 +- sound/soc/codecs/wm8776.c | 2 +- sound/soc/codecs/wm8900.c | 2 +- sound/soc/codecs/wm8940.c | 2 +- sound/soc/codecs/wm8961.c | 2 +- sound/soc/codecs/wm8988.c | 2 +- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm9081.c | 2 +- sound/soc/codecs/wm9705.c | 2 +- sound/soc/codecs/wm9712.c | 2 +- sound/soc/codecs/wm9713.c | 2 +- 17 files changed, 17 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 910c62a..ad5987e 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -907,7 +907,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8580 = { .probe = wm8580_probe, .remove = wm8580_remove, .set_bias_level = wm8580_set_bias_level, - .reg_cache_size = sizeof(wm8580_reg), + .reg_cache_size = ARRAY_SIZE(wm8580_reg), .reg_word_size = sizeof(u16), .reg_cache_default = &wm8580_reg, }; diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index f8d9c60..52a923a 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -418,7 +418,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8711 = { .suspend = wm8711_suspend, .resume = wm8711_resume, .set_bias_level = wm8711_set_bias_level, - .reg_cache_size = sizeof(wm8711_reg), + .reg_cache_size = ARRAY_SIZE(wm8711_reg), .reg_word_size = sizeof(u16), .reg_cache_default = wm8711_reg, }; diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index 5e5a162..32eba85 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -272,7 +272,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8728 = { .suspend = wm8728_suspend, .resume = wm8728_resume, .set_bias_level = wm8728_set_bias_level, - .reg_cache_size = sizeof(wm8728_reg_defaults), + .reg_cache_size = ARRAY_SIZE(wm8728_reg_defaults), .reg_word_size = sizeof(u16), .reg_cache_default = wm8728_reg_defaults, }; diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 56f5408..2827501 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -567,7 +567,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8731 = { .suspend = wm8731_suspend, .resume = wm8731_resume, .set_bias_level = wm8731_set_bias_level, - .reg_cache_size = sizeof(wm8731_reg), + .reg_cache_size = ARRAY_SIZE(wm8731_reg), .reg_word_size = sizeof(u16), .reg_cache_default = wm8731_reg, }; diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index c8e7a26..90e31e9 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -453,7 +453,7 @@ static int wm8741_probe(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_wm8741 = { .probe = wm8741_probe, .resume = wm8741_resume, - .reg_cache_size = sizeof(wm8741_reg_defaults), + .reg_cache_size = ARRAY_SIZE(wm8741_reg_defaults), .reg_word_size = sizeof(u16), .reg_cache_default = &wm8741_reg_defaults, }; diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 92d4b2a..4d1ec9d 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -747,7 +747,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8750 = { .suspend = wm8750_suspend, .resume = wm8750_resume, .set_bias_level = wm8750_set_bias_level, - .reg_cache_size = sizeof(wm8750_reg), + .reg_cache_size = ARRAY_SIZE(wm8750_reg), .reg_word_size = sizeof(u16), .reg_cache_default = wm8750_reg, }; diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 4a945d3..0b0e3b3 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1617,7 +1617,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8753 = { .suspend = wm8753_suspend, .resume = wm8753_resume, .set_bias_level = wm8753_set_bias_level, - .reg_cache_size = sizeof(wm8753_reg), + .reg_cache_size = ARRAY_SIZE(wm8753_reg), .reg_word_size = sizeof(u16), .reg_cache_default = wm8753_reg, }; diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index 05d98dd..c0b6917 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -448,7 +448,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8776 = { .suspend = wm8776_suspend, .resume = wm8776_resume, .set_bias_level = wm8776_set_bias_level, - .reg_cache_size = sizeof(wm8776_reg), + .reg_cache_size = ARRAY_SIZE(wm8776_reg), .reg_word_size = sizeof(u16), .reg_cache_default = wm8776_reg, }; diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 1378aab..25c8ccd 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -1256,7 +1256,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8900 = { .resume = wm8900_resume, .set_bias_level = wm8900_set_bias_level, .volatile_register = wm8900_volatile_register, - .reg_cache_size = sizeof(wm8900_reg_defaults), + .reg_cache_size = ARRAY_SIZE(wm8900_reg_defaults), .reg_word_size = sizeof(u16), .reg_cache_default = wm8900_reg_defaults, }; diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index d28bf0d..2cb16f8 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -750,7 +750,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8940 = { .suspend = wm8940_suspend, .resume = wm8940_resume, .set_bias_level = wm8940_set_bias_level, - .reg_cache_size = sizeof(wm8940_reg_defaults), + .reg_cache_size = ARRAY_SIZE(wm8940_reg_defaults), .reg_word_size = sizeof(u16), .reg_cache_default = wm8940_reg_defaults, }; diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index bfc6578..4f326f6 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -1075,7 +1075,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8961 = { .suspend = wm8961_suspend, .resume = wm8961_resume, .set_bias_level = wm8961_set_bias_level, - .reg_cache_size = sizeof(wm8961_reg_defaults), + .reg_cache_size = ARRAY_SIZE(wm8961_reg_defaults), .reg_word_size = sizeof(u16), .reg_cache_default = wm8961_reg_defaults, .volatile_register = wm8961_volatile_register, diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index d070a58..d08b1fe 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -809,7 +809,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8988 = { .suspend = wm8988_suspend, .resume = wm8988_resume, .set_bias_level = wm8988_set_bias_level, - .reg_cache_size = sizeof(wm8988_reg), + .reg_cache_size = ARRAY_SIZE(wm8988_reg), .reg_word_size = sizeof(u16), .reg_cache_default = wm8988_reg, }; diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index a3fd6b9..589e3fa 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1586,7 +1586,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8993 = { .suspend = wm8993_suspend, .resume = wm8993_resume, .set_bias_level = wm8993_set_bias_level, - .reg_cache_size = sizeof(wm8993_reg_defaults), + .reg_cache_size = ARRAY_SIZE(wm8993_reg_defaults), .reg_word_size = sizeof(u16), .reg_cache_default = wm8993_reg_defaults, .volatile_register = wm8993_volatile, diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 00249d5..ecc7c37 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1317,7 +1317,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm9081 = { .suspend = wm9081_suspend, .resume = wm9081_resume, .set_bias_level = wm9081_set_bias_level, - .reg_cache_size = sizeof(wm9081_reg_defaults), + .reg_cache_size = ARRAY_SIZE(wm9081_reg_defaults), .reg_word_size = sizeof(u16), .reg_cache_default = wm9081_reg_defaults, .volatile_register = wm9081_volatile_register, diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index e4d8f53..a144acd 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -385,7 +385,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm9705 = { .resume = wm9705_soc_resume, .read = ac97_read, .write = ac97_write, - .reg_cache_size = sizeof(wm9705_reg), + .reg_cache_size = ARRAY_SIZE(wm9705_reg), .reg_word_size = sizeof(u16), .reg_cache_step = 2, .reg_cache_default = wm9705_reg, diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index cebeb93..d2f224d 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -674,7 +674,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm9712 = { .read = ac97_read, .write = ac97_write, .set_bias_level = wm9712_set_bias_level, - .reg_cache_size = sizeof(wm9712_reg), + .reg_cache_size = ARRAY_SIZE(wm9712_reg), .reg_word_size = sizeof(u16), .reg_cache_step = 2, .reg_cache_default = wm9712_reg, diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 463917e..7da13b0 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1257,7 +1257,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm9713 = { .read = ac97_read, .write = ac97_write, .set_bias_level = wm9713_set_bias_level, - .reg_cache_size = sizeof(wm9713_reg), + .reg_cache_size = ARRAY_SIZE(wm9713_reg), .reg_word_size = sizeof(u16), .reg_cache_step = 2, .reg_cache_default = wm9713_reg, -- 1.7.2.3

3 2

[alsa-devel] [PATCH] ALSA: usb-audio: Assume first control interface is for audio
by Daniel Mack 10 Sep '10

10 Sep '10

For devices with more than one control interface, let's assume the first one contains the audio controls. Unfortunately, there is no field in any of the descriptors to tell us whether a control interface is for audio or MIDI controls, so a better check is not easy to implement. On a composite device with audio and MIDI functions, for example, the code currently overwrites chip->ctrl_intf, causing operations on the control interface to fail if they are issued after the device probe. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/usb/card.c | 8 +++++++- 1 files changed, 7 insertions(+), 1 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 9feb00c..b443a33 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -465,7 +465,13 @@ static void *snd_usb_audio_probe(struct usb_device *dev, goto __error; } - chip->ctrl_intf = alts; + /* + * For devices with more than one control interface, we assume the + * first contains the audio controls. We might need a more specific + * check here in the future. + */ + if (!chip->ctrl_intf) + chip->ctrl_intf = alts; if (err > 0) { /* create normal USB audio interfaces */ -- 1.7.0.4

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[alsa-devel] [PATCH - 1/1] patch_nvhdmi.c: Add missing codec IDs, unify names
by Stephen Warren 10 Sep '10

10 Sep '10

* Add missing codec IDs. * Modify some existing codec names for discrete GPUs to match newly added IDs. Note: existing names were a mixture of marketing and engineering GPU names. Equally, there's no reason that codec IDs have to be specific to a particular GPU or board, so identify codecs in a less marketing-oriented fashion. * Reformat codec ID table so it's easier to read, for me at least. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> diff --git a/pci/hda/patch_nvhdmi.c b/pci/hda/patch_nvhdmi.c index a281836..77e2b40 100644 --- a/pci/hda/patch_nvhdmi.c +++ b/pci/hda/patch_nvhdmi.c @@ -540,26 +540,32 @@ static int patch_nvhdmi_2ch(struct hda_codec *codec) * patch entries */ static struct hda_codec_preset snd_hda_preset_nvhdmi[] = { - { .id = 0x10de0002, .name = "MCP77/78 HDMI", - .patch = patch_nvhdmi_8ch_7x }, - { .id = 0x10de0003, .name = "MCP77/78 HDMI", - .patch = patch_nvhdmi_8ch_7x }, - { .id = 0x10de0005, .name = "MCP77/78 HDMI", - .patch = patch_nvhdmi_8ch_7x }, - { .id = 0x10de0006, .name = "MCP77/78 HDMI", - .patch = patch_nvhdmi_8ch_7x }, - { .id = 0x10de0007, .name = "MCP79/7A HDMI", - .patch = patch_nvhdmi_8ch_7x }, - { .id = 0x10de000a, .name = "GT220 HDMI", - .patch = patch_nvhdmi_8ch_89 }, - { .id = 0x10de000b, .name = "GT21x HDMI", - .patch = patch_nvhdmi_8ch_89 }, - { .id = 0x10de000c, .name = "MCP89 HDMI", - .patch = patch_nvhdmi_8ch_89 }, - { .id = 0x10de000d, .name = "GT240 HDMI", - .patch = patch_nvhdmi_8ch_89 }, - { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch }, - { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch }, + { .id = 0x10de0002, .name = "MCP77/78 HDMI", .patch = patch_nvhdmi_8ch_7x }, + { .id = 0x10de0003, .name = "MCP77/78 HDMI", .patch = patch_nvhdmi_8ch_7x }, + { .id = 0x10de0005, .name = "MCP77/78 HDMI", .patch = patch_nvhdmi_8ch_7x }, + { .id = 0x10de0006, .name = "MCP77/78 HDMI", .patch = patch_nvhdmi_8ch_7x }, + { .id = 0x10de0007, .name = "MCP79/7A HDMI", .patch = patch_nvhdmi_8ch_7x }, + { .id = 0x10de000a, .name = "GPU 0a HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de000b, .name = "GPU 0b HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de000c, .name = "MCP89 HDMI", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de000d, .name = "GPU 0d HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0010, .name = "GPU 10 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0011, .name = "GPU 11 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0012, .name = "GPU 12 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0013, .name = "GPU 13 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0014, .name = "GPU 14 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0018, .name = "GPU 18 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0019, .name = "GPU 19 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de001a, .name = "GPU 1a HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de001b, .name = "GPU 1b HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de001c, .name = "GPU 1c HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0040, .name = "GPU 40 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0041, .name = "GPU 41 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0042, .name = "GPU 42 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0043, .name = "GPU 43 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0044, .name = "GPU 44 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch }, + { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch }, {} /* terminator */ }; @@ -572,6 +578,21 @@ MODULE_ALIAS("snd-hda-codec-id:10de000a"); MODULE_ALIAS("snd-hda-codec-id:10de000b"); MODULE_ALIAS("snd-hda-codec-id:10de000c"); MODULE_ALIAS("snd-hda-codec-id:10de000d"); +MODULE_ALIAS("snd-hda-codec-id:10de0010"); +MODULE_ALIAS("snd-hda-codec-id:10de0011"); +MODULE_ALIAS("snd-hda-codec-id:10de0012"); +MODULE_ALIAS("snd-hda-codec-id:10de0013"); +MODULE_ALIAS("snd-hda-codec-id:10de0014"); +MODULE_ALIAS("snd-hda-codec-id:10de0018"); +MODULE_ALIAS("snd-hda-codec-id:10de0019"); +MODULE_ALIAS("snd-hda-codec-id:10de001a"); +MODULE_ALIAS("snd-hda-codec-id:10de001b"); +MODULE_ALIAS("snd-hda-codec-id:10de001c"); +MODULE_ALIAS("snd-hda-codec-id:10de0040"); +MODULE_ALIAS("snd-hda-codec-id:10de0041"); +MODULE_ALIAS("snd-hda-codec-id:10de0042"); +MODULE_ALIAS("snd-hda-codec-id:10de0043"); +MODULE_ALIAS("snd-hda-codec-id:10de0044"); MODULE_ALIAS("snd-hda-codec-id:10de0067"); MODULE_ALIAS("snd-hda-codec-id:10de8001"); -- 1.6.0.6 nvpublic

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[alsa-devel] [PATCH 1/1] mfd: add codec resource into 88pm860x driver
by Haojian Zhuang 10 Sep '10

10 Sep '10

3 2

[alsa-devel] [PATCH 0/5 v2] fbdev: sh_mobile_hdmi: bug fix patches V2
by Kuninori Morimoto 10 Sep '10

10 Sep '10

Dear Mark, Liam, Guennadi These are ALSA V2 bug fix patches which are reported by Guennadi. Kuninori Morimoto (5): fbdev: sh_mobile_hdmi: modify noisy comment out fbdev: sh_mobile_hdmi: modify flags name to more specific fbdev: sh_mobile_hdmi: modify snd_soc_dai_driver settings fbdev: sh_mobile_hdmi: add new label for sound error path ASoC: fsi-hdmi: remove unneeded header To Mark. Please care above patch independently from these patches. [alsa-devel] [PATCH 5/5] ASoC: fsi-ak4642: modiry platform_name These series are for ALSA side patches. I will send SH side V2 patches soon. Main diff v1 <==> v2 is I added Guennadi's report mail to log area on each patches, and care more. But these series still didn't care above. I added reasons. -- Guennadi ------------------------------------------ > +config SND_FSI_HDMI > + bool "FSI-HDMI sound support" > + depends on SND_SOC_SH4_FSI && FB_SH_MOBILE_HDMI > + help > + This option enables generic sound support for the > + FSI - HDMI unit > + "bool" means, if someone is linking the whole ASoC into the kernel, they will not be able to build this as a module. Not a big deal, but you're stealing some freedom from the user. ------------------------------------------------------- understand. But the config which use "bool" for FSI-XXX is not only FSI-HDMI. So, I will care your indication as "new function patch" in future. -- Guennadi ------------------------------------------ With this config option you will have 3 SND_SOC_SH4_FSI implementations in the Kconfig, all selectable independently. Do you really think it makes sense and would work, if someone were to select more than one of those options at the same time? ------------------------------------------------------- Yes. I created it for independently. For example, you can select FSI-AK4642 and FSI-HDMI in same time, and both works well for me on AP4 board now. But I guess, if you select FSI-DA7210 and FSI-HDMI, small patch which change FSIA <-> FSIB is needed. FSI-AK4642 : FSI-A FSI-DA7210 : FSI-B FSI-HDMI : FSI-B I think that it should be going to enable the selection of "FSI-A" and "FSI-B" in the future. But now, no way, no idea. It is my task By the way, about HDMI sound, I know that there are some HDMI monitors which can not use sound. Now I'm debuging about it. -- Guennadi ------------------------------------------ > + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, Again, this I am not sure about. The datasheet says 16 to 32 bit are possible, but then I only see configuration for 16 to 24 bits, but in any case, I think, you'd want to implement .hw_params to support non-default formats. ------------------------------------------------------- Yes. I should modify this .formats. But I can not select, test, support 32bit in current environment. So, please put it to my TODO list. And yes. .hw_params implementation is important for advanced support. But now, current HDMI sound support is still prototype (I didn't indicate though). So. please put it to my TODO list too. Best regards -- Kuninori Morimoto

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[alsa-devel] [PATCH] ASoC: fsi codecs: modify menu attribute on Kconfig
by Kuninori Morimoto 10 Sep '10

10 Sep '10

Current SND_FSI_xxx menu attributes were bool, but it should be tristate. This patch solve below report from Guennadi "bool" means, if someone is linking the whole ASoC into the kernel, they will not be able to build this as a module. Not a big deal, but you're stealing some freedom from the user. Reported-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/Kconfig | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 6b224d2..7f0a496 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -47,7 +47,7 @@ config SND_SH7760_AC97 AC97 unit of the SH7760. config SND_FSI_AK4642 - bool "FSI-AK4642 sound support" + tristate "FSI-AK4642 sound support" depends on SND_SOC_SH4_FSI && I2C_SH_MOBILE select SND_SOC_AK4642 help @@ -55,7 +55,7 @@ config SND_FSI_AK4642 FSI - AK4642 unit config SND_FSI_DA7210 - bool "FSI-DA7210 sound support" + tristate "FSI-DA7210 sound support" depends on SND_SOC_SH4_FSI && I2C_SH_MOBILE select SND_SOC_DA7210 help @@ -63,7 +63,7 @@ config SND_FSI_DA7210 FSI - DA7210 unit config SND_FSI_HDMI - bool "FSI-HDMI sound support" + tristate "FSI-HDMI sound support" depends on SND_SOC_SH4_FSI && FB_SH_MOBILE_HDMI help This option enables generic sound support for the -- 1.7.0.4

3 2

[alsa-devel] [PATCH 3/3] ASoC: Samsung: Debug PCM platform device name
by Jassi Brar 10 Sep '10

10 Sep '10

From: Jassi Brar <jassi.brar(a)samsung.com> The PCM controller platform devices are registered by the name 'samsung-pcm', so use the same in the CPU driver. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/s3c-pcm.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/s3c24xx/s3c-pcm.c b/sound/soc/s3c24xx/s3c-pcm.c index d404f80..6e9139f 100644 --- a/sound/soc/s3c24xx/s3c-pcm.c +++ b/sound/soc/s3c24xx/s3c-pcm.c @@ -525,7 +525,7 @@ static struct platform_driver s3c_pcm_driver = { .probe = s3c_pcm_dev_probe, .remove = s3c_pcm_dev_remove, .driver = { - .name = "samsung-pcm-audio", + .name = "samsung-pcm", .owner = THIS_MODULE, }, }; @@ -546,4 +546,4 @@ module_exit(s3c_pcm_exit); MODULE_AUTHOR("Jaswinder Singh, <jassi.brar(a)samsung.com>"); MODULE_DESCRIPTION("S3C PCM Controller Driver"); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:samsung-pcm-audio"); +MODULE_ALIAS("platform:samsung-pcm"); -- 1.6.2.5

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[alsa-devel] [PATCH 1/3] ASoC: S3C: AC97: Remove the -dai suffix
by Jassi Brar 10 Sep '10

10 Sep '10

From: Jassi Brar <jassi.brar(a)samsung.com> Drop the invalid -dai suffix appended to the Samsung AC97 CPU DAI. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/smdk2443_wm9710.c | 2 +- sound/soc/s3c24xx/smdk_wm9713.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/s3c24xx/smdk2443_wm9710.c b/sound/soc/s3c24xx/smdk2443_wm9710.c index 66f9e22..4613288 100644 --- a/sound/soc/s3c24xx/smdk2443_wm9710.c +++ b/sound/soc/s3c24xx/smdk2443_wm9710.c @@ -28,7 +28,7 @@ static struct snd_soc_dai_link smdk2443_dai[] = { { .name = "AC97", .stream_name = "AC97 HiFi", - .cpu_dai_name = "s3c-ac97-dai", + .cpu_dai_name = "s3c-ac97", .codec_dai_name = "ac97-hifi", .codec_name = "ac97-codec", .platform_name = "s3c24xx-pcm-audio", diff --git a/sound/soc/s3c24xx/smdk_wm9713.c b/sound/soc/s3c24xx/smdk_wm9713.c index 90108a7..11086f8 100644 --- a/sound/soc/s3c24xx/smdk_wm9713.c +++ b/sound/soc/s3c24xx/smdk_wm9713.c @@ -47,7 +47,7 @@ static struct snd_soc_dai_link smdk_dai = { .name = "AC97", .stream_name = "AC97 PCM", .platform_name = "s3c24xx-pcm-audio", - .cpu_dai_name = "s3c-ac97-dai", + .cpu_dai_name = "s3c-ac97", .codec_dai_name = "wm9713-hifi", .codec_name = "wm9713-codec", }; -- 1.6.2.5

3 2

[alsa-devel] [PATCH 0/5] ASoC: sh_mobile_hdmi: bug fix patches
by Kuninori Morimoto 10 Sep '10

10 Sep '10

Dear Mark, Liam, Guennadi These are bug fix patches for sh_mobile_hdmi. Kuninori Morimoto (5): fbdev: sh_mobile_hdmi: modify HDMI flags name to more specific fbdev: sh_mobile_hdmi: modify snd_soc_dai_driver settings fbdev: sh_mobile_hdmi: add new label for sound error path ASoC: fsi-hdmi: remove unneeded header ASoC: fsi-ak4642: modiry platform_name 1/5 - 4/5 bugs are reported by Guennadi. And 5/5 is needed if Paul's genesis tree are merged to Mark's "for-2.6.37" Best regards -- Kuninori Morimoto

4 14

[alsa-devel] M-Audio Fast Track Ultra (8r) mixer: How to verify dB information?
by Felix Homann 10 Sep '10

10 Sep '10

Hi list, after some struggle I managed to get the mixer in the M-Audio working. Before I submit a patch I would like to know if there's an way to test the dB information for the controls. I'm using the mixer_vol_tlv TLV callback right now but I'm not sure it produces correct data. Kind regards, Felix

2 3

[alsa-devel] [PATCH 1/2] ALSA: snd-usb-caiaq: drop version number
by Daniel Mack 10 Sep '10

10 Sep '10

Let git do the job. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/usb/caiaq/device.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index cdfb856..da9cb6d 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -36,7 +36,7 @@ #include "input.h" MODULE_AUTHOR("Daniel Mack <daniel(a)caiaq.de>"); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.21"); +MODULE_DESCRIPTION("caiaq USB audio"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," -- 1.7.0.4

2 5

[alsa-devel] [PATCH 1/3] ALSA: snd-usb-caiaq: drop version number
by Daniel Mack 10 Sep '10

10 Sep '10

Let git do the job. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/usb/caiaq/device.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index cdfb856..da9cb6d 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -36,7 +36,7 @@ #include "input.h" MODULE_AUTHOR("Daniel Mack <daniel(a)caiaq.de>"); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.21"); +MODULE_DESCRIPTION("caiaq USB audio"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," -- 1.7.0.4

2 4

[alsa-devel] cmipci surround sound front-left plays on lfe
by Tom Corner 10 Sep '10

10 Sep '10

I've been going around and around on this. The surround sound 5.1 is playing fine except channel 0 (front-left) is playing on the woofer. I have a Terratec Aureon fun - CMI8738-MC6. I can redirect the output on 0 to right-front successfully using .asoundrc and "ttable.0.1 1". I have gotten desperate and have tested this card in Windows and was able to get it working right there. I have been using the file www_lynnemusic_com_surround_test.wma through mplayer and speaker-test to to test the setup. I am using pulseaudio, but I reconfigured my system without pulseaudio and get the same result. Any help would be much appreciated. Tom Corner

2 1

[alsa-devel] [PATCH] ALSA: hda - fix cirrus dac/beep coefs setup
by Jan Nikitenko 10 Sep '10

10 Sep '10

Fix copy&paste bug in cirrus coefs setup, that caused bogus value in DAC_CFG that should have gone to BEEP_CFG. The problem could be noticed with 'Headphone' output getting audible volume ramp in the beginning of sound playback. Signed-off-by: Jan Nikitenko <jan.nikitenko(a)gmail.com> --- sound/pci/hda/patch_cirrus.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 350ee8a..7d5c074 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -966,7 +966,7 @@ static struct hda_verb cs_coef_init_verbs[] = { | 0x0400 /* Disable Coefficient Auto increment */ )}, /* Beep */ - {0x11, AC_VERB_SET_COEF_INDEX, IDX_DAC_CFG}, + {0x11, AC_VERB_SET_COEF_INDEX, IDX_BEEP_CFG}, {0x11, AC_VERB_SET_PROC_COEF, 0x0007}, /* Enable Beep thru DAC1/2/3 */ {} /* terminator */ -- 1.7.1.1

1 0

[alsa-devel] [ANN] Linux.conf.au 2011 multimedia miniconf: call for papers
by Jonathan Woithe 09 Sep '10

09 Sep '10

Continuing the tradition from past conferences, the LCA Multimedia Miniconf is back at Linux.conf.au 2011 (LCA2011). Linux.conf.au 2011 will be hosted in Brisbane, Australia from 24-29 January 2011 with the miniconfs being held over the first two days. For more information about the conference check out http://lca2011.linux.org.au/. Multimedia encapsulates a wide variety of topics: composition and playpack of video, audio and music. Submissions are now invited from those interested in giving presentations or short tutorials at the Multimedia miniconf on any topic related to opensource multimedia software, such as the following: * exciting new additions to existing software * development of new multimedia programs * how to set Linux up for multimedia composition, authoring and/or playback * great user experiences with open source multimedia software * legal issues surrounding multimedia content and its use with open source software * descriptions of multimedia workflows: tools used, process followed, and so forth * upcoming features in low-level infrastructure (JACK, ALSA, PulseAudio, etc) * any other open source multimedia topic we haven't thought of It is expected that most of the miniconf presentation slots will be around 20 minutes with question time to follow. We may be able to accommodate longer 50 minute presentations depending on the number of proposals offered. In addition, time will be allocated for 5 minute "lightning talks". Projection facilities will be available. Please note that in line with linux.conf.au guidelines miniconfs cannot receive funding, nor can they provide travel or accommodation assistance. In general presenters and attendees at miniconfs must be registered for LCA2011 although miniconf-only registration can be arranged - contact the multimedia minconf organisers for more information. To submit a proposal please send the following to lca-mm(a)lists.annodex.net or one of the organisers by Thursday 6 October 2010 00:00 UT: * your name and preferred email address * the title of your talk * a brief abstract describing what you want to talk about (up to about 500 words) * your preferred presentation length For more information about the miniconf please refer to the website http://www.annodex.org/events/lca2011_mmm/ or contact one of the organisers (Jonathan Woithe, Silvia Pfeiffer, Roderick Dornan - addresses are on the website). Regards Jonathan Woithe LCA2011 Multimedia miniconf organiser

1 0

[alsa-devel] [patch] ALSA: rawmidi: cleanup the get next midi device ioctl
by Dan Carpenter 09 Sep '10

09 Sep '10

I'm doing an audit to find integer overflows and my static checker complained that in the original code "device + 1" could overflow. The overflow is harmless, but it's still worth cleaning up. The other thing that I noticed is that if you pass in a device which is higher than SNDRV_RAWMIDI_DEVICES then it doesn't return an error code but just tells you that the next device is "device + 1". I have rewritten it to just return -EINVAL if you pass in a bogus value that's either too high or too low. Signed-off-by: Dan Carpenter <error27(a)gmail.com> diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index eb68326..f944180 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -829,8 +829,12 @@ static int snd_rawmidi_control_ioctl(struct snd_card *card, if (get_user(device, (int __user *)argp)) return -EFAULT; + if (device < 0) + return -EINVAL; + if (device > SNDRV_RAWMIDI_DEVICES) + return -EINVAL; mutex_lock(&register_mutex); - device = device < 0 ? 0 : device + 1; + device++; while (device < SNDRV_RAWMIDI_DEVICES) { if (snd_rawmidi_search(card, device)) break;

5 11

[alsa-devel] cmipci surround sound front-left plays on lfe
by Tom Corner 09 Sep '10

09 Sep '10

I've been going around and around on this. The surround sound 5.1 is playing fine except channel 0 (front-left) is playing on the woofer. I have a Terratec Aureon fun - CMI8738-MC6. I can redirect the output on 0 to right-front successfully using .asoundrc and "ttable.0.1 1". I have gotten desperate and have tested this card in Windows and was able to get it working right there. I have been using the file www_lynnemusic_com_surround_test.wma through mplayer and speaker-test to to test the setup. I am using pulseaudio, but I reconfigured my system without pulseaudio and get the same result. Any help would be much appreciated. Tom Corner

1 0

[alsa-devel] USB Audio device syncpipe error
by 6275 kson 09 Sep '10

09 Sep '10

Hello, I am working on a usb audio device and I'm having trouble getting it to show up as an output device in Ubuntu's sound preferences. The device is working well with other OS's. My syslog shows this error. 'cannot submit syncpipe for urb 0, error -90: internal error' If someone could help me diagnose the source of the problem I would appreciate it very much. Here are some more details. The device operates at high speed but uses USB Audio Class 1.0. The output endpoint is Isochronous asynchronous and I am using a feedback endpoint with a 32ms polling interval. The device works with Ubuntu when running at full speed. (Input and Output show up in the sound preferences). At High Speed only the Input shows up in the sound preferences. _From my usb analyzer log I never see a request for the feedback endpoint when running at HS, while I do see the request at FS. Cheers,

3 3

[alsa-devel] [PATCH] ALSA: HDA: Add fixup pins for Ideapad Y550
by David Henningsson 09 Sep '10

09 Sep '10

By adding the subwoofer as a speaker pin, it is treated correctly when auto-muting. I'm a little unsure of whether it should go to stable as well, because it needs yesterday's commit bb35febd to make sense (in fact, I developed parts of that patch in parallel). Any opinions? -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 1

[PATCH] ALSA: HDA: Add fixup pins for Ideapad Y550
by David Henningsson 09 Sep '10

09 Sep '10

By adding the subwoofer as a speaker pin, it is treated correctly when auto-muting. BugLink: https://launchpad.net/bugs/611803 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 26 +++++++++++++++++++++++++- 1 files changed, 25 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f11a9ca..0c25d22 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -18963,6 +18963,26 @@ static void alc662_auto_init(struct hda_codec *codec) alc_inithook(codec); } +enum { + ALC662_FIXUP_IDEAPAD, +}; + +static const struct alc_fixup alc662_fixups[] = { + [ALC662_FIXUP_IDEAPAD] = { + .pins = (const struct alc_pincfg[]) { + { 0x17, 0x99130112 }, /* subwoofer */ + { } + } + }, +}; + +static struct snd_pci_quirk alc662_fixup_tbl[] = { + SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD), + {} +}; + + + static int patch_alc662(struct hda_codec *codec) { struct alc_spec *spec; @@ -18995,6 +19015,7 @@ static int patch_alc662(struct hda_codec *codec) } if (board_config == ALC662_AUTO) { + alc_pick_fixup(codec, alc662_fixup_tbl, alc662_fixups, 1); /* automatic parse from the BIOS config */ err = alc662_parse_auto_config(codec); if (err < 0) { @@ -19053,8 +19074,11 @@ static int patch_alc662(struct hda_codec *codec) spec->vmaster_nid = 0x02; codec->patch_ops = alc_patch_ops; - if (board_config == ALC662_AUTO) + if (board_config == ALC662_AUTO) { spec->init_hook = alc662_auto_init; + alc_pick_fixup(codec, alc662_fixup_tbl, alc662_fixups, 0); + } + #ifdef CONFIG_SND_HDA_POWER_SAVE if (!spec->loopback.amplist) spec->loopback.amplist = alc662_loopbacks; -- 1.7.1 --------------090907070406080900090002--

1 0

[PATCH] ALSA: hda - Clear left-over hp_pins in
by Takashi Iwai 08 Sep '10

08 Sep '10

snd_hda_parse_pin_def_config() In snd_hda_parse_def_config(), some unused values may remain in hp_pins[] array during the headphone-reassignment workaround. This patch clears the unused array members. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 2980c27..bfdde7b 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4558,6 +4558,8 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, memmove(sequences_hp + i, sequences_hp + i + 1, sizeof(sequences_hp[0]) * (cfg->hp_outs - i)); } + memset(cfg->hp_pins + cfg->hp_outs, 0, + sizeof(hda_nid_t) * (AUTO_CFG_MAX_OUTS - cfg->hp_outs)); } /* sort by sequence */ -- 1.7.2.2 ================ -- ~Nirbheek Chauhan Gentoo GNOME+Mozilla Team

1 0

[alsa-devel] [PATCH] OMAP: McBSP: Fix static function warning
by G, Manjunath Kondaiah 08 Sep '10

08 Sep '10

This patch fixes sparse warning due non declaration of static function sound/soc/omap/omap-mcbsp.c:783:5: warning: symbol 'omap_mcbsp_st_info_volsw' was not declared. Should it be static? Signed-off-by: G, Manjunath Kondaiah <manjugk(a)ti.com> Cc: alsa-devel(a)alsa-project.org Cc: Jarkko Nikula <jhnikula(a)gmail.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> Cc: Liam Girdwood <lrg(a)slimlogic.co.uk> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Tony Lindgren <tony(a)atomide.com> Cc: Nishanth Menon <nm(a)ti.com> --- sound/soc/omap/omap-mcbsp.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 86f2139..5768381 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -780,7 +780,7 @@ struct snd_soc_dai omap_mcbsp_dai[] = { EXPORT_SYMBOL_GPL(omap_mcbsp_dai); -int omap_mcbsp_st_info_volsw(struct snd_kcontrol *kcontrol, +static int omap_mcbsp_st_info_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct soc_mixer_control *mc = -- 1.7.0.4

5 4

[alsa-devel] [PATCH 0/4] ASoC: rx1950-related fixes
by Vasily Khoruzhick 08 Sep '10

08 Sep '10

Some cosmetical fixes for rx1950 and uda1380 ASoC drivers

2 6

[alsa-devel] [patch 0/3] ASoC: t5325 audio support
by Arnaud Patard 08 Sep '10

08 Sep '10

Hi, This patchset is the first try to get merged audio support for HP t5325 thinclient. This board is using a ALC5621 so the major part of this series is made of the ALC5623 driver. It's supporting ALC5621/22/23 but has been tested only with a ALC5621. Arnaud

2 8

[alsa-devel] [patch] sound/oss: potential integer overflow
by Dan Carpenter 08 Sep '10

08 Sep '10

We don't want "pre_event_timeout" to be negative because that would result in a stack traces in dmesg when we schedule a negative timeout. In the original code "HZ * val" could overflow so I just moved the check for negative below the multiply. Also the code in snd_seq_oss_ioctl() deliberately set the timeout to -1 which is wrong. Signed-off-by: Dan Carpenter <error27(a)gmail.com> diff --git a/sound/oss/sequencer.c b/sound/oss/sequencer.c index e85789e..de61295 100644 --- a/sound/oss/sequencer.c +++ b/sound/oss/sequencer.c @@ -1507,9 +1507,9 @@ int sequencer_ioctl(int dev, struct file *file, unsigned int cmd, void __user *a case SNDCTL_MIDI_PRETIME: if (get_user(val, p)) return -EFAULT; + val = (HZ * val) / 10; if (val < 0) val = 0; - val = (HZ * val) / 10; pre_event_timeout = val; break; diff --git a/sound/oss/midibuf.c b/sound/oss/midibuf.c index 782b3b8..73acacd 100644 --- a/sound/oss/midibuf.c +++ b/sound/oss/midibuf.c @@ -380,9 +380,9 @@ int MIDIbuf_ioctl(int dev, struct file *file, case SNDCTL_MIDI_PRETIME: if (get_user(val, (int __user *)arg)) return -EFAULT; + val = (HZ * val) / 10; if (val < 0) val = 0; - val = (HZ * val) / 10; parms[dev].prech_timeout = val; return put_user(val, (int __user *)arg); diff --git a/sound/core/seq/oss/seq_oss_ioctl.c b/sound/core/seq/oss/seq_oss_ioctl.c index 5ac701c..ae2a39f 100644 --- a/sound/core/seq/oss/seq_oss_ioctl.c +++ b/sound/core/seq/oss/seq_oss_ioctl.c @@ -191,10 +191,9 @@ snd_seq_oss_ioctl(struct seq_oss_devinfo *dp, unsigned int cmd, unsigned long ca return 0; if (get_user(val, p)) return -EFAULT; - if (val <= 0) - val = -1; - else - val = (HZ * val) / 10; + val = (HZ * val) / 10; + if (val < 0) + val = 0; dp->readq->pre_event_timeout = val; return put_user(val, p) ? -EFAULT : 0;

3 2