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[alsa-devel] [PATCH] pcm: use the same set of uint_XX types everywhere
by John Spencer 11 Oct '13

11 Oct '13

previously, pcm.h prototyped the functions snd_pcm_format_silence* with u_int_XX types (legacy BSD types from sys/types.h), while the implementation in pcm_misc.c used the C99 stdint.h types uint_XX. this caused compilation errors on systems where the typedef for u_int_XX and uint_XX is not exactly the same. we fix it by using the C99 stdint.h types everywhere. Signed-off-by: John Spencer <maillist-alsa(a)barfooze.de> --- include/pcm.h | 9 +++++---- src/pcm/pcm_misc.c | 42 +++++++++++++++++++++--------------------- 2 files changed, 26 insertions(+), 25 deletions(-) diff --git a/include/pcm.h b/include/pcm.h index 95b8aed..cd847e4 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -33,6 +33,7 @@ extern "C" { #endif +#include <stdint.h> /** * \defgroup PCM PCM Interface * See the \ref pcm page for more details. @@ -1048,10 +1049,10 @@ int snd_pcm_format_width(snd_pcm_format_t format); /* in bits */ int snd_pcm_format_physical_width(snd_pcm_format_t format); /* in bits */ snd_pcm_format_t snd_pcm_build_linear_format(int width, int pwidth, int unsignd, int big_endian); ssize_t snd_pcm_format_size(snd_pcm_format_t format, size_t samples); -u_int8_t snd_pcm_format_silence(snd_pcm_format_t format); -u_int16_t snd_pcm_format_silence_16(snd_pcm_format_t format); -u_int32_t snd_pcm_format_silence_32(snd_pcm_format_t format); -u_int64_t snd_pcm_format_silence_64(snd_pcm_format_t format); +uint8_t snd_pcm_format_silence(snd_pcm_format_t format); +uint16_t snd_pcm_format_silence_16(snd_pcm_format_t format); +uint32_t snd_pcm_format_silence_32(snd_pcm_format_t format); +uint64_t snd_pcm_format_silence_64(snd_pcm_format_t format); int snd_pcm_format_set_silence(snd_pcm_format_t format, void *buf, unsigned int samples); snd_pcm_sframes_t snd_pcm_bytes_to_frames(snd_pcm_t *pcm, ssize_t bytes); diff --git a/src/pcm/pcm_misc.c b/src/pcm/pcm_misc.c index d52160c..2a8be21 100644 --- a/src/pcm/pcm_misc.c +++ b/src/pcm/pcm_misc.c @@ -363,7 +363,7 @@ ssize_t snd_pcm_format_size(snd_pcm_format_t format, size_t samples) * \param format Sample format * \return silence 64 bit word */ -u_int64_t snd_pcm_format_silence_64(snd_pcm_format_t format) +uint64_t snd_pcm_format_silence_64(snd_pcm_format_t format) { switch (format) { case SNDRV_PCM_FORMAT_S8: @@ -437,7 +437,7 @@ u_int64_t snd_pcm_format_silence_64(snd_pcm_format_t format) { union { float f[2]; - u_int64_t i; + uint64_t i; } u; u.f[0] = u.f[1] = 0.0; #ifdef SNDRV_LITTLE_ENDIAN @@ -450,7 +450,7 @@ u_int64_t snd_pcm_format_silence_64(snd_pcm_format_t format) { union { double f; - u_int64_t i; + uint64_t i; } u; u.f = 0.0; #ifdef SNDRV_LITTLE_ENDIAN @@ -463,7 +463,7 @@ u_int64_t snd_pcm_format_silence_64(snd_pcm_format_t format) { union { float f[2]; - u_int64_t i; + uint64_t i; } u; u.f[0] = u.f[1] = 0.0; #ifdef SNDRV_LITTLE_ENDIAN @@ -476,7 +476,7 @@ u_int64_t snd_pcm_format_silence_64(snd_pcm_format_t format) { union { double f; - u_int64_t i; + uint64_t i; } u; u.f = 0.0; #ifdef SNDRV_LITTLE_ENDIAN @@ -509,10 +509,10 @@ u_int64_t snd_pcm_format_silence_64(snd_pcm_format_t format) * \param format Sample format * \return silence 32 bit word */ -u_int32_t snd_pcm_format_silence_32(snd_pcm_format_t format) +uint32_t snd_pcm_format_silence_32(snd_pcm_format_t format) { assert(snd_pcm_format_physical_width(format) <= 32); - return (u_int32_t)snd_pcm_format_silence_64(format); + return (uint32_t)snd_pcm_format_silence_64(format); } /** @@ -520,10 +520,10 @@ u_int32_t snd_pcm_format_silence_32(snd_pcm_format_t format) * \param format Sample format * \return silence 16 bit word */ -u_int16_t snd_pcm_format_silence_16(snd_pcm_format_t format) +uint16_t snd_pcm_format_silence_16(snd_pcm_format_t format) { assert(snd_pcm_format_physical_width(format) <= 16); - return (u_int16_t)snd_pcm_format_silence_64(format); + return (uint16_t)snd_pcm_format_silence_64(format); } /** @@ -531,10 +531,10 @@ u_int16_t snd_pcm_format_silence_16(snd_pcm_format_t format) * \param format Sample format * \return silence 8 bit word */ -u_int8_t snd_pcm_format_silence(snd_pcm_format_t format) +uint8_t snd_pcm_format_silence(snd_pcm_format_t format) { assert(snd_pcm_format_physical_width(format) <= 8); - return (u_int8_t)snd_pcm_format_silence_64(format); + return (uint8_t)snd_pcm_format_silence_64(format); } /** @@ -550,7 +550,7 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int return 0; switch (snd_pcm_format_physical_width(format)) { case 4: { - u_int8_t silence = snd_pcm_format_silence_64(format); + uint8_t silence = snd_pcm_format_silence_64(format); unsigned int samples1; if (samples % 2 != 0) return -EINVAL; @@ -559,13 +559,13 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int break; } case 8: { - u_int8_t silence = snd_pcm_format_silence_64(format); + uint8_t silence = snd_pcm_format_silence_64(format); memset(data, silence, samples); break; } case 16: { - u_int16_t silence = snd_pcm_format_silence_64(format); - u_int16_t *pdata = (u_int16_t *)data; + uint16_t silence = snd_pcm_format_silence_64(format); + uint16_t *pdata = (uint16_t *)data; if (! silence) memset(data, 0, samples * 2); else { @@ -575,8 +575,8 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int break; } case 24: { - u_int32_t silence = snd_pcm_format_silence_64(format); - u_int8_t *pdata = (u_int8_t *)data; + uint32_t silence = snd_pcm_format_silence_64(format); + uint8_t *pdata = (uint8_t *)data; if (! silence) memset(data, 0, samples * 3); else { @@ -595,8 +595,8 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int break; } case 32: { - u_int32_t silence = snd_pcm_format_silence_64(format); - u_int32_t *pdata = (u_int32_t *)data; + uint32_t silence = snd_pcm_format_silence_64(format); + uint32_t *pdata = (uint32_t *)data; if (! silence) memset(data, 0, samples * 4); else { @@ -606,8 +606,8 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int break; } case 64: { - u_int64_t silence = snd_pcm_format_silence_64(format); - u_int64_t *pdata = (u_int64_t *)data; + uint64_t silence = snd_pcm_format_silence_64(format); + uint64_t *pdata = (uint64_t *)data; if (! silence) memset(data, 0, samples * 8); else { -- 1.7.3.4

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[alsa-devel] [PATCH] ALSA: hda - Sony VAIO Pro 13 (haswell) now has a working headset jack
by David Henningsson 11 Oct '13

11 Oct '13

Just got the positive confirmation from a tester: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1227093/comments/28 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 79e6fe7..bf313be 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3761,6 +3761,8 @@ static const struct hda_fixup alc269_fixups[] = { { 0x18, 0x01a1913c }, /* use as headset mic, without its own jack detect */ { } }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MIC }, [ALC269_FIXUP_ASUS_X101_FUNC] = { .type = HDA_FIXUP_FUNC, -- 1.7.9.5

2 1

[alsa-devel] [PATCH v2 1/2] ASoC: mc13783: add mixer controls
by Steffen Trumtrar 11 Oct '13

11 Oct '13

Add more controls to the alsa mixer infrastructure. Signed-off-by: Steffen Trumtrar <s.trumtrar(a)pengutronix.de> --- Changes in v2: - Fix some control names - remove controls that belong to DAPM sound/soc/codecs/mc13783.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index eedbf05..2b62737 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -541,8 +541,26 @@ static const struct soc_enum mc13783_enum_3d_mixer = static struct snd_kcontrol_new mc13783_control_list[] = { SOC_SINGLE("Loudspeaker enable", MC13783_AUDIO_RX0, 5, 1, 0), SOC_SINGLE("PCM Playback Volume", MC13783_AUDIO_RX1, 6, 15, 0), + SOC_SINGLE("PCM Playback Switch", MC13783_AUDIO_RX1, 5, 1, 0), SOC_DOUBLE("PCM Capture Volume", MC13783_AUDIO_TX, 19, 14, 31, 0), SOC_ENUM("3D Control", mc13783_enum_3d_mixer), + + SOC_SINGLE("CDCOUT Switch", MC13783_AUDIO_RX0, 18, 1, 0), + SOC_SINGLE("Earpiece Amp Switch", MC13783_AUDIO_RX0, 3, 1, 0), + SOC_DOUBLE("Headset Amp Switch", MC13783_AUDIO_RX0, 10, 9, 1, 0), + SOC_DOUBLE("Line out Amp Switch", MC13783_AUDIO_RX0, 16, 15, 1, 0), + + SOC_SINGLE("PCM Capture Mixin Switch", MC13783_AUDIO_RX0, 22, 1, 0), + SOC_SINGLE("Line in Capture Mixin Switch", MC13783_AUDIO_RX0, 23, 1, 0), + + SOC_SINGLE("CODEC Capture Volume", MC13783_AUDIO_RX1, 1, 15, 0), + SOC_SINGLE("CODEC Capture Mixin Switch", MC13783_AUDIO_RX0, 21, 1, 0), + + SOC_SINGLE("Line in Capture Volume", MC13783_AUDIO_RX1, 12, 15, 0), + SOC_SINGLE("Line in Capture Switch", MC13783_AUDIO_RX1, 10, 1, 0), + + SOC_SINGLE("MC1 Capture Bias Switch", MC13783_AUDIO_TX, 0, 1, 0), + SOC_SINGLE("MC2 Capture Bias Switch", MC13783_AUDIO_TX, 1, 1, 0), }; static int mc13783_probe(struct snd_soc_codec *codec) -- 1.8.4.rc3

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[alsa-devel] [PATCH v2 0/2] Sony vaio pro 13 (haswell)
by David Henningsson 11 Oct '13

11 Oct '13

Changes since v1: - Mic patch now applies to linux-stable branch linux-3.11.y cleanly - The two patches are now not connected to each other. I'll send a patch that connects them if/when we get some confirmation that this actually is a working headset mic pin. David Henningsson (2): ALSA: hda - Fix microphone for Sony VAIO Pro 13 (Haswell model) ALSA: hda - Add a headset mic model for ALC269 and friends Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_realtek.c | 24 ++++++++++++++++++++++++ 2 files changed, 25 insertions(+) -- 1.7.9.5

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[alsa-devel] [PATCH] local.h: include sys/types.h to fix issues with pcm.h
by John Spencer 11 Oct '13

11 Oct '13

sys/types.h is required for the u_int_XX types used by pcm.h. since a change in pcm.h is not desired, we add the inclusion to the header that includes pcm.h during build. Signed-off-by: John Spencer <maillist-alsa(a)barfooze.de> --- include/local.h | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/include/local.h b/include/local.h index f0db188..9464efa 100644 --- a/include/local.h +++ b/include/local.h @@ -31,6 +31,7 @@ #include <endian.h> #include <stdarg.h> #include <sys/poll.h> +#include <sys/types.h> #include <errno.h> #include <linux/types.h> #include <linux/ioctl.h> -- 1.7.3.4

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Re: [alsa-devel] Fast Track Ultra 8r...
by Felix Homann 11 Oct '13

11 Oct '13

Am 10.10.2013 19:41 schrieb "Petri Järvisalo" <petri.jarvisalo(a)gmail.com>: > > When plugging in the device at other OS, the internal routing for inputs is disabled and software routes straight 1 = 1 to 8 = 8, like this, and there is no effects applied. Those defaults are the worst I can think of for *my* use case. That's why I haven't implemented them. I do fully appreciate that those would be sane defaults for many people. I wouldn't mind anyone implementing them in Alsa. It's just, well, I won't ;-) When I started working on the FTU mixer support I didn't think about the defaults at all. The default we have right now are the ones you get without explicitly specifying any. But in fact, now I completely fell in love with these insane defaults: They give me an instant alert that I haven't restored my personal 'right' settings! That's of extreme value for my purposes as I use the device mainly on live occasions and the 'wrong' settings certainly cannot go unnoticed in the soundcheck. So again, anybody interested is invited to implement new defaults for the FTU. I won' t object any choice, how could I? But I won't do it myself since I'm personally very happy with the current state. Regards, Felix

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[alsa-devel] ALSA: HDA Control and node attachment
by Jason Steigler 10 Oct '13

10 Oct '13

Hello, I was wondering if it's possible from user-space to change which node a control is attached to. In older ALSA (1.0.23), I had used the '3stack-6ch-dig' model to get 6 channel audio for an "Realtek ALC662 rev 1" codec. I had configured the front panel mic (node 0x19) to be an output using hda-verb that could be controlled by the Center and LFE mixer controls. The board also has jacks on the back and one of those would match the audio coming out the front. When I change the volume or mute the controls, they both were affected (which I expected/wanted). With newer ALSA (1.0.24 with debian Wheezy's stock kernel), the model '3stack-6ch-dig' doesn't exist, but the codec can be configured to 6 channels. I have made the same adjustments with hda-verb to make the node an output. The volume control works for the 'front' and 'rear' jacks, but only muting works on the 'rear' jack. I noticed that the control nodes for Center and LFE are not the same between the 2 setups. In the older version, they were tied to node 0x0e (Audio Mixer), and with the newer version node 0x18 (which looks like the 'Rear Mic' jack). I am not knowledgeable of the details of the HDA codec (yet), and am mostly curious if they can be changed in user-space or if I'll have to start learning the HDA codec code and try to find a way to move the controls around. I tried using hda-jack-retask and could assign the mic nodes as outputs, but wasn't sure if it would allow me to move controls on the HDA graph. Thanks in advance! Jason

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[alsa-devel] Fast Track Ultra 8r...
by Petri Järvisalo 10 Oct '13

10 Oct '13

Hello, I'm new here and somewhat understand how to script php and python ... and very unskilled with linux... anyway i noticed that the default alsa mixer values for fast track ultra 8r are way off the optimal when the sound card is first time plugged in. What I expected was clear stereo/surround sound, what i got was distorted mono sound with reverb! Far away from optimal settings! So, I stated to seek knowledge on web, and finally stumbled upon a homepage that had some info about the Linux support. What i found out was that the sound card has internal routing system, zero latency and a effect processor.Also i found out that the routing works by controlling the input/output levels at alsa mixer. The problem, by default all alsa mixer values are assigned to 100%, this means the internal routing is routing everything to everything distorting the sound, and making the output mono! The website I found had suggestion and a bash script to correct the internal routing: http://heikki.ketoharju.info/2013/03/linux-and-fasttrackultra/#comment-989 alas, the main article is a good read too :) Well the script did not work for me, so i changed the settings manually to my alsa mixer, which took quite some time, and it works like a charm. Anyway my best wish would be to get a fix for this sound card, so it would "just work" when plugged in. Thanks for any info on how i could get this "patch" to be widely available so others could benefit it too. Best regards, Petri

2 1

[alsa-devel] [PATCH] pcm.h: include sys/types.h, necessary for u_int_XX types
by John Spencer 10 Oct '13

10 Oct '13

fixes compilation on musl libc. --- include/pcm.h | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/include/pcm.h b/include/pcm.h index 95b8aed..7ea9e84 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -29,6 +29,8 @@ #ifndef __ALSA_PCM_H #define __ALSA_PCM_H +#include <sys/types.h> + #ifdef __cplusplus extern "C" { #endif -- 1.7.3.4

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[alsa-devel] Internal Mic Boost channel is unused
by Nathanael D. Noblet 10 Oct '13

10 Oct '13

Hello, I have an issue I'd like to help completely solve. Unfortunately I only have a tenuous grasp of what is actually going on so some of this could be wrong/include errors. Here's what I know. 1) I have the following hardware (alsa-info attached). 2) The internal microphone requires that the mic boost channel be something other than 0 to function properly. 3) Changing what pulseaudio expects things to be labelled causes the problem to go away. (via the attached patch to /usr/share/pulseaudio/paths/analog-input-mic.conf). From the discussion with David Henningsson (diwic on IRC). It seems that pulse doesn't expect a Mic Boost channel to be used with internal microphones. As such to fix this particular hardware, the driver would need to make the internal mic boost be labelled "Internal Mic Boost" as opposed to Mic Boost, which is then is used for both external and internal mics. I'd love to be able to help any way I can to get these fixed once and for all. Let me know if you need more information or what not. -- Nathanael d. Noblet t 403.875.4613

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[alsa-devel] [PATCH 1/2] ALSA: hda - Add a headset mic model for ALC269 and friends
by David Henningsson 10 Oct '13

10 Oct '13

Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_realtek.c | 15 +++++++++++++++ 2 files changed, 16 insertions(+) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index f911e36..ba143cd 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -28,6 +28,7 @@ ALC269/270/275/276/28x/29x alc269-dmic Enable ALC269(VA) digital mic workaround alc271-dmic Enable ALC271X digital mic workaround inv-dmic Inverted internal mic workaround + headset-mic Indicates a combined headset (mic+headphone jack) lenovo-dock Enables docking station I/O for some Lenovos dell-headset-multi Headset jack, which can also be used as mic-in dell-headset-dock Headset jack (without mic-in), and also dock I/O diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ed9deb6..8cb3350 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2819,6 +2819,15 @@ static void alc269_fixup_hweq(struct hda_codec *codec, alc_write_coef_idx(codec, 0x1e, coef | 0x80); } +static void alc269_fixup_headset_mic(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) + spec->parse_flags |= HDA_PINCFG_HEADSET_MIC; +} + static void alc271_fixup_dmic(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -3516,6 +3525,7 @@ enum { ALC271_FIXUP_DMIC, ALC269_FIXUP_PCM_44K, ALC269_FIXUP_STEREO_DMIC, + ALC269_FIXUP_HEADSET_MIC, ALC269_FIXUP_QUANTA_MUTE, ALC269_FIXUP_LIFEBOOK, ALC269_FIXUP_AMIC, @@ -3614,6 +3624,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_stereo_dmic, }, + [ALC269_FIXUP_HEADSET_MIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_headset_mic, + }, [ALC269_FIXUP_QUANTA_MUTE] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_quanta_mute, @@ -3979,6 +3993,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC269_FIXUP_STEREO_DMIC, .name = "alc269-dmic"}, {.id = ALC271_FIXUP_DMIC, .name = "alc271-dmic"}, {.id = ALC269_FIXUP_INV_DMIC, .name = "inv-dmic"}, + {.id = ALC269_FIXUP_HEADSET_MIC, .name = "headset-mic"}, {.id = ALC269_FIXUP_LENOVO_DOCK, .name = "lenovo-dock"}, {.id = ALC269_FIXUP_HP_GPIO_LED, .name = "hp-gpio-led"}, {.id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "dell-headset-multi"}, -- 1.7.9.5

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[alsa-devel] [PATCH] ALSA: hda - Distinguish between the headset mic pin and the TRS mic
by David Henningsson 10 Oct '13

10 Oct '13

On the Acer AO722, pin 0x1a represents the mic part of the TRRS headset jack, and pin 0x1b represents the TRS mic jack. Mark pin 0x1a as headset mic will help us know which one is which. Reported-by: Nathanael D. Noblet <nathanael(a)gnat.ca> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 25 +++++++++++++++++++++++++ 1 file changed, 25 insertions(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index ec68eae..7f30c9f 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3231,6 +3231,8 @@ enum { CXT_FIXUP_INC_MIC_BOOST, CXT_FIXUP_HEADPHONE_MIC_PIN, CXT_FIXUP_HEADPHONE_MIC, + CXT_FIXUP_HEADSET_MIC_PIN, + CXT_FIXUP_HEADSET_MIC, CXT_FIXUP_GPIO1, }; @@ -3306,6 +3308,15 @@ static void cxt_fixup_headphone_mic(struct hda_codec *codec, } } +static void cxt_fixup_headset_mic(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct conexant_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) + spec->parse_flags |= HDA_PINCFG_HEADSET_MIC; +} + /* ThinkPad X200 & co with cxt5051 */ static const struct hda_pintbl cxt_pincfg_lenovo_x200[] = { @@ -3376,6 +3387,19 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = cxt_fixup_headphone_mic, }, + [CXT_FIXUP_HEADSET_MIC_PIN] = { + .type = HDA_FIXUP_PINS, + .chained = true, + .chain_id = CXT_FIXUP_HEADSET_MIC, + .v.pins = (const struct hda_pintbl[]) { + { 0x1a, 0x03a1903c }, /* use as headset mic, with jack detect */ + { } + } + }, + [CXT_FIXUP_HEADSET_MIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = cxt_fixup_headset_mic, + }, [CXT_FIXUP_GPIO1] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -3395,6 +3419,7 @@ static const struct snd_pci_quirk cxt5051_fixups[] = { static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT_FIXUP_GPIO1), + SND_PCI_QUIRK(0x1025, 0x0598, "Acer AO722", CXT_FIXUP_HEADSET_MIC_PIN), SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T410", CXT_PINCFG_LENOVO_TP410), -- 1.7.9.5

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[alsa-devel] [PATCHv2] pcm.h: include sys/types.h, necessary for u_int_XX types
by John Spencer 10 Oct '13

10 Oct '13

fixes compilation on musl libc. Signed-off-by: John Spencer <maillist-alsa(a)barfooze.de> --- include/pcm.h | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/include/pcm.h b/include/pcm.h index 95b8aed..7ea9e84 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -29,6 +29,8 @@ #ifndef __ALSA_PCM_H #define __ALSA_PCM_H +#include <sys/types.h> + #ifdef __cplusplus extern "C" { #endif -- 1.7.3.4

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[alsa-devel] [RFC] ASoC: pcm: Trigger all commands on error
by Markus Pargmann 10 Oct '13

10 Oct '13

If a start command is triggered and an error occures in the first called function, e.g. DMA, all other functions are not executed and the error value is returned immediately. In case of a start command, the calling function will call undo_start, calling this trigger function with the stop command. It will call stop for each function. But only the first function was started previously. The other functions may fail in the assumption that a stop command always comes after a start command. As the API does not specify the behaviour of trigger functionpointers, I think this should be fixed in the function calling the trigger functionpointers. This patch changes the behaviour. The trigger function calls all functionpointers independent of their returncodes. The first error-code is returned. Signed-off-by: Markus Pargmann <mpa(a)pengutronix.de> --- sound/soc/soc-pcm.c | 18 ++++++++++-------- 1 file changed, 10 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 330c9a6..23fc25b 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -616,26 +616,28 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; + int ret = 0; + int err = 0; if (codec_dai->driver->ops->trigger) { ret = codec_dai->driver->ops->trigger(substream, cmd, codec_dai); - if (ret < 0) - return ret; + if (!err && ret) + err = ret; } if (platform->driver->ops && platform->driver->ops->trigger) { ret = platform->driver->ops->trigger(substream, cmd); - if (ret < 0) - return ret; + if (!err && ret) + err = ret; } if (cpu_dai->driver->ops->trigger) { ret = cpu_dai->driver->ops->trigger(substream, cmd, cpu_dai); - if (ret < 0) - return ret; + if (!err && ret) + err = ret; } - return 0; + + return err; } static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, -- 1.8.4.rc3

2 13

[alsa-devel] [PATCH] ASoC: mc13783: add mixer controls
by Steffen Trumtrar 10 Oct '13

10 Oct '13

Add more kcontrols for the alsa mixer infrastructure. Signed-off-by: Steffen Trumtrar <s.trumtrar(a)pengutronix.de> --- sound/soc/codecs/mc13783.c | 48 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 48 insertions(+) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index eedbf05..fafe999 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -534,6 +534,30 @@ static struct snd_soc_dapm_route mc13783_routes[] = { static const char * const mc13783_3d_mixer[] = {"Stereo", "Phase Mix", "Mono", "Mono Mix"}; +static const char * const mc13783_alsp[] = {"Off", "Codec", "Right"}; + +static const char * const mc13783_ahs[] = {"Codec", "Mixer"}; + +static const struct soc_enum mc13783_enum_asp = + SOC_ENUM_SINGLE(MC13783_AUDIO_RX0, 4, ARRAY_SIZE(mc13783_alsp), + mc13783_alsp); + +static const struct soc_enum mc13783_enum_alsp = + SOC_ENUM_SINGLE(MC13783_AUDIO_RX0, 7, ARRAY_SIZE(mc13783_alsp), + mc13783_alsp); + +static const struct soc_enum mc13783_enum_ahs = + SOC_ENUM_SINGLE(MC13783_AUDIO_RX0, 11, ARRAY_SIZE(mc13783_ahs), + mc13783_ahs); + +static const struct soc_enum mc13783_enum_arxout = + SOC_ENUM_SINGLE(MC13783_AUDIO_RX0, 17, ARRAY_SIZE(mc13783_ahs), + mc13783_ahs); + +static const struct soc_enum mc13783_enum_codec = + SOC_ENUM_SINGLE(MC13783_AUDIO_RX1, 4, ARRAY_SIZE(mc13783_alsp), + mc13783_alsp); + static const struct soc_enum mc13783_enum_3d_mixer = SOC_ENUM_SINGLE(MC13783_AUDIO_RX1, 16, ARRAY_SIZE(mc13783_3d_mixer), mc13783_3d_mixer); @@ -541,8 +565,32 @@ static const struct soc_enum mc13783_enum_3d_mixer = static struct snd_kcontrol_new mc13783_control_list[] = { SOC_SINGLE("Loudspeaker enable", MC13783_AUDIO_RX0, 5, 1, 0), SOC_SINGLE("PCM Playback Volume", MC13783_AUDIO_RX1, 6, 15, 0), + SOC_SINGLE("PCM Playback Switch", MC13783_AUDIO_RX1, 5, 1, 0), SOC_DOUBLE("PCM Capture Volume", MC13783_AUDIO_TX, 19, 14, 31, 0), SOC_ENUM("3D Control", mc13783_enum_3d_mixer), + SOC_ENUM("Earpiece Switch", mc13783_enum_asp), + SOC_ENUM("Loudspeaker Switch", mc13783_enum_alsp), + + SOC_ENUM("Headset Switch", mc13783_enum_ahs), + SOC_SINGLE("Headset Amp Right Switch", MC13783_AUDIO_RX0, 9, 1, 0), + SOC_SINGLE("Headset Amp Left Switch", MC13783_AUDIO_RX0, 10, 1, 0), + + SOC_ENUM("Line out Amp Switch", mc13783_enum_arxout), + SOC_SINGLE("Line out Amp Right Switch", MC13783_AUDIO_RX0, 15, 1, 0), + SOC_SINGLE("Line out Amp Left Switch", MC13783_AUDIO_RX0, 16, 1, 0), + + SOC_SINGLE("Codec Capture Mix Switch", MC13783_AUDIO_RX0, 21, 1, 0), + SOC_SINGLE("PCM Capture Mix Switch", MC13783_AUDIO_RX0, 22, 1, 0), + SOC_SINGLE("Line in Capture Mix Switch", MC13783_AUDIO_RX0, 23, 1, 0), + + SOC_SINGLE("Codec Capture Volume", MC13783_AUDIO_RX1, 1, 15, 0), + SOC_SINGLE("Codec Capture Switch", MC13783_AUDIO_RX1, 0, 1, 0), + + SOC_SINGLE("PGA Capture Volume", MC13783_AUDIO_RX1, 12, 15, 0), + SOC_SINGLE("PGA Capture Switch", MC13783_AUDIO_RX1, 10, 1, 0), + + SOC_SINGLE("MC1 Capture Bias Switch", MC13783_AUDIO_TX, 0, 1, 0), + SOC_SINGLE("MC2 Capture Bias Switch", MC13783_AUDIO_TX, 1, 1, 0), }; static int mc13783_probe(struct snd_soc_codec *codec) -- 1.8.4.rc3

3 5

[alsa-devel] [PATCH] ALSA: hda - Enable surround speakers (when line out is also present)
by David Henningsson 10 Oct '13

10 Oct '13

In the case where we have both line out and more than stereo speakers, the speaker DACs will end up in extra_out_nid. In fact, AFAIU, speakers are the only ones that can end up in extra_out_nid, and if we have several of those, they should be surround outputs rather than copy front. BugLink: https://bugs.launchpad.net/bugs/1236965 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_codec.c | 19 ++++++++++++++----- 1 file changed, 14 insertions(+), 5 deletions(-) Notes: * Thanks to Raymond Yau for the pointer * Not tested (yet), except in hda-emu * Not sure if this should be sent to stable or not, given that it is a change of behaviour * Alsainfo available here: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1236965/+attachm… diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 5b6c4e3..68801ba 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5395,11 +5395,6 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, snd_hda_codec_setup_stream(codec, mout->hp_out_nid[i], stream_tag, 0, format); - for (i = 0; i < ARRAY_SIZE(mout->extra_out_nid); i++) - if (!mout->no_share_stream && mout->extra_out_nid[i]) - snd_hda_codec_setup_stream(codec, - mout->extra_out_nid[i], - stream_tag, 0, format); /* surrounds */ for (i = 1; i < mout->num_dacs; i++) { @@ -5410,6 +5405,20 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, snd_hda_codec_setup_stream(codec, nids[i], stream_tag, 0, format); } + + /* extra surrounds */ + for (i = 0; i < ARRAY_SIZE(mout->extra_out_nid); i++) { + int ch = 0; + if (!mout->extra_out_nid[i]) + break; + if (chs >= (i + 1) * 2) + ch = i * 2; + else if (!mout->no_share_stream) + break; + snd_hda_codec_setup_stream(codec, mout->extra_out_nid[i], + stream_tag, ch, format); + } + return 0; } EXPORT_SYMBOL_HDA(snd_hda_multi_out_analog_prepare); -- 1.7.9.5

2 1

Re: [alsa-devel] Cannot load snd-usb-audio on 3.12-rc2
by Takashi Iwai 10 Oct '13

10 Oct '13

At Mon, 23 Sep 2013 16:41:07 -0700, Sarah Sharp wrote: > > The snd-usb-audio driver no longer loads properly on 3.12-rc2 when I > plug in my USB headset. It worked fine on 3.11-rc4 (not sure about > vanilla 3.11). > > Trying to manually load the driver fails: > > sarah@xanatos:~$ sudo modprobe snd-usb-audio > FATAL: Error inserting snd_usb_audio > (/lib/modules/3.12.0-rc2/kernel/sound/usb/snd-usb-audio.ko): Invalid argument > > dmesg shows: > > [ 44.231668] snd_usb_audio: `-2' invalid for parameter `index' There is no corresponding code change in the USB audio driver side, so I guess it must be in the module parameter parser that got broken. Does the patch below fix? thanks, Takashi --- diff --git a/kernel/params.c b/kernel/params.c index 81c4e78..c00d5b5 100644 --- a/kernel/params.c +++ b/kernel/params.c @@ -254,11 +254,11 @@ int parse_args(const char *doing, STANDARD_PARAM_DEF(byte, unsigned char, "%hhu", unsigned long, kstrtoul); -STANDARD_PARAM_DEF(short, short, "%hi", long, kstrtoul); +STANDARD_PARAM_DEF(short, short, "%hi", long, kstrtol); STANDARD_PARAM_DEF(ushort, unsigned short, "%hu", unsigned long, kstrtoul); -STANDARD_PARAM_DEF(int, int, "%i", long, kstrtoul); +STANDARD_PARAM_DEF(int, int, "%i", long, kstrtol); STANDARD_PARAM_DEF(uint, unsigned int, "%u", unsigned long, kstrtoul); -STANDARD_PARAM_DEF(long, long, "%li", long, kstrtoul); +STANDARD_PARAM_DEF(long, long, "%li", long, kstrtol); STANDARD_PARAM_DEF(ulong, unsigned long, "%lu", unsigned long, kstrtoul); int param_set_charp(const char *val, const struct kernel_param *kp)

2 2

[alsa-devel] firewire amdtp details
by Uli Franke 09 Oct '13

09 Oct '13

Hi Clemens One of the reasons why we are working on snd_dice is because we experienced (rare but unacceptable nevertheless) audio dropouts with the original driver. The DICE PLL lost lock in such cases. Investigations showed that this varies with the hardware/temperature/sun spots/snowfall. This led me to the conclusion that the device buffering/SYT-timing must be somehow close to its limits (late/early). In such a case just little node clock differences (induced by temperature and alike) can cause the system to drop out of the PLL's locking range. During capture mode SYT implementation/review I re-read the SYT code and found the following: from amdtp_stream_set_parameters: /* default buffering in the device */ s->transfer_delay = TRANSFER_DELAY_TICKS - TICKS_PER_CYCLE; if (s->flags & CIP_BLOCKING) /* additional buffering needed to adjust for no-data packets */ s->transfer_delay += TICKS_PER_SECOND * amdtp_syt_intervals[sfc] / rate; which sets a global transfer delay. But later in calculate_syt I get the impression that (not exactly) the same operations are performed again accidentally: if (syt_offset < TICKS_PER_CYCLE) { syt_offset += TRANSFER_DELAY_TICKS - TICKS_PER_CYCLE; if (s->flags & CIP_BLOCKING) syt_offset += s->transfer_delay; syt = (cycle + syt_offset / TICKS_PER_CYCLE) << 12; syt += syt_offset % TICKS_PER_CYCLE; return syt & 0xffff; } When I interpret the code from set_parameters as some sort of precomputation, the code in calculate_syt should be something like this: if (syt_offset < TICKS_PER_CYCLE) { syt_offset += s->transfer_delay; syt = (cycle + syt_offset / TICKS_PER_CYCLE) << 12; syt += syt_offset % TICKS_PER_CYCLE; return syt & 0xffff; } This could explain that the stream gets delayed too much - resulting in late timing (to be verified though). Therefore I'm a bit curious if you could enlighten me about this as s->transfer_delay isn't used in any other place. Additionally I appreciate any justifications for the subtraction of TICKS_PER_CYCLE from the global transfer delay. This would help me to get a better understanding of snd_dice's SYT mechanics. Regards Uli

2 1

[alsa-devel] [PATCH] ASoC: dapm: Use async I/O for DAPM sequences
by Mark Brown 09 Oct '13

09 Oct '13

From: Mark Brown <broonie(a)linaro.org> Within a DAPM sequence we normally don't care about when exactly a register write has completed so long as they happen in the order we requested. This means that we can issue most of the writes we do asynchronously which should maximise the ability of the underlying frameworks to keep the hardware busy, providing a small performance improvement on some systems. We currently ensure that all writes are completed both when changing to a different device and when calling into the regulator and clock frameworks. This should ensure that the previous ordering is maintained. We also ensure that writes are completed prior to calling into widget event functions since some event functions implement delays. This should be improved in future so that widgets can disable this sync in order to add extra writes. Signed-off-by: Mark Brown <broonie(a)linaro.org> xvdfvsd --- sound/soc/soc-dapm.c | 24 ++++++++++++++++++++++-- 1 file changed, 22 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 25b9554..905e4c5 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -409,6 +409,12 @@ static inline void soc_widget_unlock(struct snd_soc_dapm_widget *w) mutex_unlock(&w->platform->mutex); } +static void soc_dapm_async_complete(struct snd_soc_dapm_context *dapm) +{ + if (dapm->codec && dapm->codec->using_regmap) + regmap_async_complete(dapm->codec->control_data); +} + static int soc_widget_update_bits_locked(struct snd_soc_dapm_widget *w, unsigned short reg, unsigned int mask, unsigned int value) { @@ -417,8 +423,9 @@ static int soc_widget_update_bits_locked(struct snd_soc_dapm_widget *w, int ret; if (w->codec && w->codec->using_regmap) { - ret = regmap_update_bits_check(w->codec->control_data, - reg, mask, value, &change); + ret = regmap_update_bits_check_async(w->codec->control_data, + reg, mask, value, + &change); if (ret != 0) return ret; } else { @@ -1201,6 +1208,8 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, { int ret; + soc_dapm_async_complete(w->dapm); + if (SND_SOC_DAPM_EVENT_ON(event)) { if (w->on_val & SND_SOC_DAPM_REGULATOR_BYPASS) { ret = regulator_allow_bypass(w->regulator, false); @@ -1234,6 +1243,8 @@ int dapm_clock_event(struct snd_soc_dapm_widget *w, if (!w->clk) return -EIO; + soc_dapm_async_complete(w->dapm); + #ifdef CONFIG_HAVE_CLK if (SND_SOC_DAPM_EVENT_ON(event)) { return clk_prepare_enable(w->clk); @@ -1426,6 +1437,7 @@ static void dapm_seq_check_event(struct snd_soc_card *card, if (w->event && (w->event_flags & event)) { pop_dbg(w->dapm->dev, card->pop_time, "pop test : %s %s\n", w->name, ev_name); + soc_dapm_async_complete(w->dapm); trace_snd_soc_dapm_widget_event_start(w, event); ret = w->event(w, NULL, event); trace_snd_soc_dapm_widget_event_done(w, event); @@ -1498,6 +1510,7 @@ static void dapm_seq_run(struct snd_soc_card *card, struct list_head *list, int event, bool power_up) { struct snd_soc_dapm_widget *w, *n; + struct snd_soc_dapm_context *d; LIST_HEAD(pending); int cur_sort = -1; int cur_subseq = -1; @@ -1528,6 +1541,9 @@ static void dapm_seq_run(struct snd_soc_card *card, cur_subseq); } + if (cur_dapm && w->dapm != cur_dapm) + soc_dapm_async_complete(cur_dapm); + INIT_LIST_HEAD(&pending); cur_sort = -1; cur_subseq = INT_MIN; @@ -1586,6 +1602,10 @@ static void dapm_seq_run(struct snd_soc_card *card, cur_dapm->seq_notifier(cur_dapm, i, cur_subseq); } + + list_for_each_entry(d, &card->dapm_list, list) { + soc_dapm_async_complete(d); + } } static void dapm_widget_update(struct snd_soc_card *card) -- 1.8.4.rc3

1 0

[alsa-devel] [PATCH] ASoC: dapm: Run clock and regulator events separately to other supplies
by Mark Brown 09 Oct '13

09 Oct '13

From: Mark Brown <broonie(a)linaro.org> In order to avoid trying to use an external clock or supply for an on-chip supply prior to it being enabled move the clock and regulator supply events to a separate step in DAPM sequencing from normal supply events. This should have minimal practical impact since these widgets are sorted using SND_SOC_NOPM which is a negative value and hence sorted separately to any real register writes, though it may be relevant if supplies have event callbacks only. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/soc-dapm.c | 50 +++++++++++++++++++++++++------------------------- 1 file changed, 25 insertions(+), 25 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 905e4c5..eb26d5a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -59,31 +59,31 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, /* dapm power sequences - make this per codec in the future */ static int dapm_up_seq[] = { [snd_soc_dapm_pre] = 0, - [snd_soc_dapm_supply] = 1, [snd_soc_dapm_regulator_supply] = 1, [snd_soc_dapm_clock_supply] = 1, - [snd_soc_dapm_micbias] = 2, + [snd_soc_dapm_supply] = 2, + [snd_soc_dapm_micbias] = 3, [snd_soc_dapm_dai_link] = 2, - [snd_soc_dapm_dai_in] = 3, - [snd_soc_dapm_dai_out] = 3, - [snd_soc_dapm_aif_in] = 3, - [snd_soc_dapm_aif_out] = 3, - [snd_soc_dapm_mic] = 4, - [snd_soc_dapm_mux] = 5, - [snd_soc_dapm_virt_mux] = 5, - [snd_soc_dapm_value_mux] = 5, - [snd_soc_dapm_dac] = 6, - [snd_soc_dapm_switch] = 7, - [snd_soc_dapm_mixer] = 7, - [snd_soc_dapm_mixer_named_ctl] = 7, - [snd_soc_dapm_pga] = 8, - [snd_soc_dapm_adc] = 9, - [snd_soc_dapm_out_drv] = 10, - [snd_soc_dapm_hp] = 10, - [snd_soc_dapm_spk] = 10, - [snd_soc_dapm_line] = 10, - [snd_soc_dapm_kcontrol] = 11, - [snd_soc_dapm_post] = 12, + [snd_soc_dapm_dai_in] = 4, + [snd_soc_dapm_dai_out] = 4, + [snd_soc_dapm_aif_in] = 4, + [snd_soc_dapm_aif_out] = 4, + [snd_soc_dapm_mic] = 5, + [snd_soc_dapm_mux] = 6, + [snd_soc_dapm_virt_mux] = 6, + [snd_soc_dapm_value_mux] = 6, + [snd_soc_dapm_dac] = 7, + [snd_soc_dapm_switch] = 8, + [snd_soc_dapm_mixer] = 8, + [snd_soc_dapm_mixer_named_ctl] = 8, + [snd_soc_dapm_pga] = 9, + [snd_soc_dapm_adc] = 10, + [snd_soc_dapm_out_drv] = 11, + [snd_soc_dapm_hp] = 11, + [snd_soc_dapm_spk] = 11, + [snd_soc_dapm_line] = 11, + [snd_soc_dapm_kcontrol] = 12, + [snd_soc_dapm_post] = 13, }; static int dapm_down_seq[] = { @@ -109,10 +109,10 @@ static int dapm_down_seq[] = { [snd_soc_dapm_dai_in] = 10, [snd_soc_dapm_dai_out] = 10, [snd_soc_dapm_dai_link] = 11, - [snd_soc_dapm_clock_supply] = 12, - [snd_soc_dapm_regulator_supply] = 12, [snd_soc_dapm_supply] = 12, - [snd_soc_dapm_post] = 13, + [snd_soc_dapm_clock_supply] = 13, + [snd_soc_dapm_regulator_supply] = 13, + [snd_soc_dapm_post] = 14, }; static void pop_wait(u32 pop_time) -- 1.8.4.rc3

1 0

[alsa-devel] [PATCH 1/1] ALSA: usb-audio: Use module_usb_driver
by Sachin Kamat 09 Oct '13

09 Oct '13

module_usb_driver makes code simpler by removing the boilerplate. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/usb/card.c | 13 +------------ 1 file changed, 1 insertion(+), 12 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 9d9de8d..d979050 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -751,15 +751,4 @@ static struct usb_driver usb_audio_driver = { .supports_autosuspend = 1, }; -static int __init snd_usb_audio_init(void) -{ - return usb_register(&usb_audio_driver); -} - -static void __exit snd_usb_audio_cleanup(void) -{ - usb_deregister(&usb_audio_driver); -} - -module_init(snd_usb_audio_init); -module_exit(snd_usb_audio_cleanup); +module_usb_driver(usb_audio_driver); -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: hda - Add fixup for ASUS N56VZ
by Takashi Iwai 08 Oct '13

08 Oct '13

ASUS N56VZ needs a fixup for the bass speaker pin, which was already provided via model=asus-mode4. Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=841645 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 52c26d3..ed9deb6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4596,6 +4596,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), + SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_ASUS_MODE4), SND_PCI_QUIRK(0x1043, 0x8469, "ASUS mobo", ALC662_FIXUP_NO_JACK_DETECT), SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_FIXUP_ASUS_MODE2), SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD), -- 1.8.4

1 0

Re: [alsa-devel] [Alsa-user] hdmi sound rerouted?
by Clemens Ladisch 08 Oct '13

08 Oct '13

Dagg Stompler wrote: >> daggs(a)gmx.com wrote: >>> card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] >>> card 1: Generic [HD-Audio Generic], device 3: HDMI 0 [HDMI 0] >>> card 2: HDMI [HDA ATI HDMI], device 3: HDMI 0 [HDMI 0] >> >> The second one is called "Generic" because your kernel does not (yet) >> know the hardware. >> >>> 01:00.1 Audio device: Advanced Micro Devices, Inc. [AMD/ATI] Cedar HDMI Audio [Radeon HD 5400/6300 Series] >>> 06:00.1 Audio device: Advanced Micro Devices, Inc. [AMD/ATI] RV635 HDMI Audio [Radeon HD 3600 Series] >> >> These are the HDMI outputs of *two* graphics cards. The first one's >> model is so new that the driver doesn't know it. > > the 3600 doesn't have hdmi socket so I think it is irrelevant for my issue. The RV635 chip does support HDMI outputs (even if none are actually connected), so the corresponding PCI device shows up here. >>> If you update the kernel, you will get two devices named "HDA ATI HDMI". > > upgrade to where? Oops. Even the latest driver lacks the PCI IDs of many HDMI HDA devices (aa50/aa58/aa60/aa68/aa80/aa88/aa90/aa98: Cypress/Juniper/Redwood/Cedar/ Cayman+Antilles/Barts/Turks/Caicos). Alex, are these devices that require disabling snooping? Regards, Clemens

4 5

[alsa-devel] ALSA: snd-usb-usx2y: remove bogus frame checks
by Guido Aulisi 08 Oct '13

08 Oct '13

Hi, I did some more testing on my US122: I used my patch (see below) and it worked (with default nrpack=4), but I got "should not be here with counts=42" errors, then after some time the kernel crashed, but I couldn't copy the trace. Then I set nrpack=1 and I got no "should not be here..." and it was more stable, no crash. I only checked playback, not recording, nor MIDI, with aplay I played some songs for about one hour... I hope this can be useful. But I think there are other bugs in this driver. Ciao Guido Aulisi The patch is: diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 63fb521..cd16fcc 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -328,7 +328,7 @@ static void i_usX2Y_urb_complete(struct urb *urb) usX2Y_error_urb_status(usX2Y, subs, urb); return; } - if (likely((urb->start_frame & 0xFFFF) == (usX2Y->wait_iso_frame & 0xFFFF))) + if (likely((urb->start_frame & 0x03FF) == (usX2Y->wait_iso_frame & 0x03FF))) subs->completed_urb = urb; else { usX2Y_error_sequence(usX2Y, subs, urb);

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[alsa-devel] Can't get capturing from RME Hammerfall MADI card to work: sound loops & silcence
by Niels Laukens 08 Oct '13

08 Oct '13

Hi, I'm trying to capture audio from an RME Hammerfall MADI card (PCI ID 10ee:3fc6). The setup previously worked fine on Windows, so I'm currently assuming that the card and the input signal is OK. I wrote a very simple ALSA-application that simply opens the device, sets the access format to NONINTERLEAVED (which seems to be the only option the card likes), selects 32bit sampling (although the audio is 24bit, ALSA rejects this option), 48kHz sampling and 64 channels (also, the only option the card accepts). Next, it loops with snd_pcm_readn() to get 100ms worth of audio, and dumps channel 10 to file. (Full source in attachement) The result is strange for various reasons: * The second half of every buffer returned by snd_pcm_readn() is always zero's. * The call seems to return way to fast: I would expect to get 48000 frames/second, but judging from the timing of the snd_pcm_readn() call, it's more like 1536000 frames/second (i.e. 32 fold faster). * When playing back the resulting file, I do recognize some music, but it sounds like it's the same part over and over again. Also, restarting the capture program minutes later, returns the same (at least as far as my ears & brain can recall) fragment of audio. What am I doing wrong here? (Samples of the PCM-file are available upon request) I've also tried a related test with `arecord`, with similar but different results: * The audio was also interspersed with silence * The capture duration did not match the requested '-d' option: the recording was a factor of 2 shorter than requested. * There was more audio in the recording: if I manually cut out the silent parts, the audio sounded reasonable. What am I missing here? Is this the right place to ask? Or should this go to the RME forum? Thanks in advance, Niels

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[alsa-devel] Audio Miniconference 2013 schedule - Edinburgh 21st October
by Mark Brown 07 Oct '13

07 Oct '13

The audio mini-summit in Edinburgh is now less than a month away so it seems like a good time to start pulling together a schedule. Looking at the topics people mentioned when filling out the registration form there were three major topics that people wanted to cover: - Compressed audio review: tinycompress, integration into userspace, driver upstreaming. - UCM and other use case management status & review. - Effects offload to DSPs with dynamically instantiatable algorithms, and general dynamic DSP firmware handling. Plus a few things that only one or two people brought up: - Process stuff (the list, bug tracking and so on). - Some relatively specific hardware support topics: - HDMI hotplug handling throughout the stack - Mic status LED handling throughout the stack. - Code cleanups: - Moving away from the custom ALSA device registration infrastructure. - General code modernisation. - General routing control in userspace - UIs, automation. The first question is if there is anything important that is missing from the above list - please speak up soon if there is! It'd also be good to have volunteers to lead the discussions in each area. In terms of format I'd propose that we go through the list above in roughly that order, having someone lead discussion on the topic. Does this sound good to people? As previously mentioned it'd be good if anyone planning to attend could fill in the form at: https://docs.google.com/forms/d/105uav9tkwwkDU4P1T3IOWjfsa3kgtDbFAJOlHgBQR5… to help with planning & coordination.

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[alsa-devel] [PATCH v2] ALSA: snd-usb-usx2y: remove bogus frame checks
by Daniel Mack 07 Oct '13

07 Oct '13

The frame check in i_usX2Y_urb_complete() and i_usX2Y_usbpcm_urb_complete() is bogus and produces false positives as described in this LAU thread: http://linuxaudio.org/mailarchive/lau/2013/5/20/200177 This patch removes the check code entirely. Cc: fzu(a)wemgehoertderstaat.de Reported-by: Dr Nicholas J Bailey <nicholas.bailey(a)glasgow.ac.uk> Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/usb/usx2y/usbusx2yaudio.c | 22 +++------------------- sound/usb/usx2y/usx2yhwdeppcm.c | 7 +------ 2 files changed, 4 insertions(+), 25 deletions(-) diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 63fb521..6234a51 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -299,19 +299,6 @@ static void usX2Y_error_urb_status(struct usX2Ydev *usX2Y, usX2Y_clients_stop(usX2Y); } -static void usX2Y_error_sequence(struct usX2Ydev *usX2Y, - struct snd_usX2Y_substream *subs, struct urb *urb) -{ - snd_printk(KERN_ERR -"Sequence Error!(hcd_frame=%i ep=%i%s;wait=%i,frame=%i).\n" -"Most probably some urb of usb-frame %i is still missing.\n" -"Cause could be too long delays in usb-hcd interrupt handling.\n", - usb_get_current_frame_number(usX2Y->dev), - subs->endpoint, usb_pipein(urb->pipe) ? "in" : "out", - usX2Y->wait_iso_frame, urb->start_frame, usX2Y->wait_iso_frame); - usX2Y_clients_stop(usX2Y); -} - static void i_usX2Y_urb_complete(struct urb *urb) { struct snd_usX2Y_substream *subs = urb->context; @@ -328,12 +315,9 @@ static void i_usX2Y_urb_complete(struct urb *urb) usX2Y_error_urb_status(usX2Y, subs, urb); return; } - if (likely((urb->start_frame & 0xFFFF) == (usX2Y->wait_iso_frame & 0xFFFF))) - subs->completed_urb = urb; - else { - usX2Y_error_sequence(usX2Y, subs, urb); - return; - } + + subs->completed_urb = urb; + { struct snd_usX2Y_substream *capsubs = usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE], *playbacksubs = usX2Y->subs[SNDRV_PCM_STREAM_PLAYBACK]; diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c index f2a1acd..814d0e8 100644 --- a/sound/usb/usx2y/usx2yhwdeppcm.c +++ b/sound/usb/usx2y/usx2yhwdeppcm.c @@ -244,13 +244,8 @@ static void i_usX2Y_usbpcm_urb_complete(struct urb *urb) usX2Y_error_urb_status(usX2Y, subs, urb); return; } - if (likely((urb->start_frame & 0xFFFF) == (usX2Y->wait_iso_frame & 0xFFFF))) - subs->completed_urb = urb; - else { - usX2Y_error_sequence(usX2Y, subs, urb); - return; - } + subs->completed_urb = urb; capsubs = usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE]; capsubs2 = usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE + 2]; playbacksubs = usX2Y->subs[SNDRV_PCM_STREAM_PLAYBACK]; -- 1.8.3.1

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[alsa-devel] [PATCH] ASoC: tlv320aic3x: Connect 'Left Line1R Mux' and 'Right Line1L Mux'
by Peter Ujfalusi 07 Oct '13

07 Oct '13

The two paths were not connected in the DAPM route causing the associated routes to be non working and the following warnings printed in the logs: tlv320aic3x-codec 0-001b: ASoC: mux Right Line1L Mux has no paths tlv320aic3x-codec 0-001b: ASoC: mux Left Line1R Mux has no paths Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/tlv320aic3x.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 892c108..a160b7d 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -678,6 +678,8 @@ static const struct snd_soc_dapm_route intercon[] = { /* Left Input */ {"Left Line1L Mux", "single-ended", "LINE1L"}, {"Left Line1L Mux", "differential", "LINE1L"}, + {"Left Line1R Mux", "single-ended", "LINE1R"}, + {"Left Line1R Mux", "differential", "LINE1R"}, {"Left Line2L Mux", "single-ended", "LINE2L"}, {"Left Line2L Mux", "differential", "LINE2L"}, @@ -694,6 +696,8 @@ static const struct snd_soc_dapm_route intercon[] = { /* Right Input */ {"Right Line1R Mux", "single-ended", "LINE1R"}, {"Right Line1R Mux", "differential", "LINE1R"}, + {"Right Line1L Mux", "single-ended", "LINE1L"}, + {"Right Line1L Mux", "differential", "LINE1L"}, {"Right Line2R Mux", "single-ended", "LINE2R"}, {"Right Line2R Mux", "differential", "LINE2R"}, -- 1.8.3.2

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Re: [alsa-devel] [PATCH] sound: usb: support wireless devices in snd_usb_parse_datainterval
by Takashi Iwai 07 Oct '13

07 Oct '13

At Tue, 1 Oct 2013 14:32:57 -0500, Thomas Pugliese wrote: > > Hi, > This patch adds support for dev speed USB_SPEED_WIRELESS in > snd_usb_parse_datainterval which allows the usb sound core to create > ISO urbs with the correct number and size of buffers. > > Signed-off-by: Thomas Pugliese <thomas.pugliese(a)gmail.com> Thanks, applied this one, too. Takashi > --- > sound/usb/helper.c | 1 + > 1 file changed, 1 insertion(+) > > diff --git a/sound/usb/helper.c b/sound/usb/helper.c > index 6209024..51ed1ac 100644 > --- a/sound/usb/helper.c > +++ b/sound/usb/helper.c > @@ -118,6 +118,7 @@ unsigned char snd_usb_parse_datainterval(struct snd_usb_audio *chip, > { > switch (snd_usb_get_speed(chip->dev)) { > case USB_SPEED_HIGH: > + case USB_SPEED_WIRELESS: > case USB_SPEED_SUPER: > if (get_endpoint(alts, 0)->bInterval >= 1 && > get_endpoint(alts, 0)->bInterval <= 4) > -- > 1.7.10.4 > > -- > To unsubscribe from this list: send the line "unsubscribe linux-usb" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html >

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Re: [alsa-devel] [PATCH] sound: usb: add support for wireless USB devices
by Takashi Iwai 07 Oct '13

07 Oct '13

At Tue, 1 Oct 2013 14:32:15 -0500, Thomas Pugliese wrote: > > This patch updates snd_usb_audio_create also support devices whose > speed == USB_SPEED_WIRELESS. > > Signed-off-by: Thomas Pugliese <thomas.pugliese(a)gmail.com> Thanks, applied now. (BTW, please add the maintainers explicitly to Cc at the next time.) Takashi > --- > sound/usb/card.c | 1 + > 1 file changed, 1 insertion(+) > > diff --git a/sound/usb/card.c b/sound/usb/card.c > index 64952e2..ef76aa0 100644 > --- a/sound/usb/card.c > +++ b/sound/usb/card.c > @@ -349,6 +349,7 @@ static int snd_usb_audio_create(struct usb_device *dev, int idx, > case USB_SPEED_LOW: > case USB_SPEED_FULL: > case USB_SPEED_HIGH: > + case USB_SPEED_WIRELESS: > case USB_SPEED_SUPER: > break; > default: > -- > 1.7.10.4 > > -- > To unsubscribe from this list: send the line "unsubscribe linux-usb" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html >

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[alsa-devel] [PATCH 1/3] ASoC: dapm: Add support for virtual mixer controls
by Lars-Peter Clausen 07 Oct '13

07 Oct '13

This patch adds support for virtual DAPM mixer controls. They are similar to virtual DAPM enums. There is no hardware register backing the control, so changing the control's value wont have any direct effect on the hardware. But it still influences the DAPM graph by causing the path it sits on to be connected or disconnected. This in turn can cause power changes for some of the widgets on the DAPM graph, which will then modify the hardware state. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc-dapm.h | 4 ++++ include/sound/soc.h | 3 ++- sound/soc/soc-dapm.c | 45 +++++++++++++++++++++++++++------------------ 3 files changed, 33 insertions(+), 19 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 27a72d5..2037c45 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -286,6 +286,8 @@ struct device; .info = snd_soc_info_volsw, \ .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 1) } +#define SOC_DAPM_SINGLE_VIRT(xname, max) \ + SOC_DAPM_SINGLE(xname, SND_SOC_NOPM, 0, max, 0) #define SOC_DAPM_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, \ @@ -300,6 +302,8 @@ struct device; .tlv.p = (tlv_array), \ .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 0) } +#define SOC_DAPM_SINGLE_TLV_VIRT(xname, max, tlv_array) \ + SOC_DAPM_SINGLE(xname, SND_SOC_NOPM, 0, max, 0, tlv_array) #define SOC_DAPM_ENUM(xname, xenum) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_enum_double, \ diff --git a/include/sound/soc.h b/include/sound/soc.h index 6dd329a..af1678c 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1049,7 +1049,8 @@ struct snd_soc_pcm_runtime { /* mixer control */ struct soc_mixer_control { int min, max, platform_max; - unsigned int reg, rreg, shift, rshift; + int reg, rreg; + unsigned int shift, rshift; unsigned int invert:1; unsigned int autodisable:1; }; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c17c14c..25b9554 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -499,18 +499,22 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, int val; struct soc_mixer_control *mc = (struct soc_mixer_control *) w->kcontrol_news[i].private_value; - unsigned int reg = mc->reg; + int reg = mc->reg; unsigned int shift = mc->shift; int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; - val = soc_widget_read(w, reg); - val = (val >> shift) & mask; - if (invert) - val = max - val; + if (reg != SND_SOC_NOPM) { + val = soc_widget_read(w, reg); + val = (val >> shift) & mask; + if (invert) + val = max - val; + p->connect = !!val; + } else { + p->connect = 0; + } - p->connect = !!val; } break; case snd_soc_dapm_mux: { @@ -2791,7 +2795,7 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, struct snd_soc_card *card = codec->card; struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; + int reg = mc->reg; unsigned int shift = mc->shift; int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; @@ -2804,7 +2808,7 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, kcontrol->id.name); mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - if (dapm_kcontrol_is_powered(kcontrol)) + if (dapm_kcontrol_is_powered(kcontrol) && reg != SND_SOC_NOPM) val = (snd_soc_read(codec, reg) >> shift) & mask; else val = dapm_kcontrol_get_value(kcontrol); @@ -2835,7 +2839,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, struct snd_soc_card *card = codec->card; struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; + int reg = mc->reg; unsigned int shift = mc->shift; int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; @@ -2857,19 +2861,24 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - dapm_kcontrol_set_value(kcontrol, val); + change = dapm_kcontrol_set_value(kcontrol, val); - mask = mask << shift; - val = val << shift; + if (reg != SND_SOC_NOPM) { + mask = mask << shift; + val = val << shift; + + change = snd_soc_test_bits(codec, reg, mask, val); + } - change = snd_soc_test_bits(codec, reg, mask, val); if (change) { - update.kcontrol = kcontrol; - update.reg = reg; - update.mask = mask; - update.val = val; + if (reg != SND_SOC_NOPM) { + update.kcontrol = kcontrol; + update.reg = reg; + update.mask = mask; + update.val = val; - card->update = &update; + card->update = &update; + } soc_dapm_mixer_update_power(card, kcontrol, connect); -- 1.8.0

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[alsa-devel] [PATCH 0/7] ALSA: hda - hdmi: Various channel mapping fixes
by Anssi Hannula 07 Oct '13

07 Oct '13

Hi! Apparently there were several more bugs in the HDMI channel mapping code, so here you go. I haven't marked any for stable since these don't really affect the regular layouts that much and the patches aren't that trivial. If you want to pick some for stable, my suggestions are 1, 5, 6, maybe 3 (those I initially marked for stable until I changed my mind). Anssi Hannula (7): ALSA: hda - hdmi: Fix reported channel map on common default layouts ALSA: hda - hdmi: Fix incorrect default channel mapping for unusual CAs ALSA: hda - hdmi: Fix programmed active channel count ALSA: hda - hdmi: Fix unused slots being enabled in manual and non-PCM mappings ALSA: hda - hdmi: Fix channel maps with less common speakers ALSA: hda - hdmi: Fix available channel maps missing from TLV ALSA: hda - hdmi: Tweak debug messages to be more useful sound/pci/hda/patch_hdmi.c | 146 +++++++++++++++++++++-------------- 1 file changed, 90 insertions(+), 56 deletions(-) -- Anssi Hannula

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[alsa-devel] [PATCH] ASoC: dapm: Ignore VMID widgets for target bias
by Lars-Peter Clausen 07 Oct '13

07 Oct '13

VMID widgets behave very similar to signal generator widgets. Both are always considered to be powered up. This means that we need to ignore the VMID widgets in the same way as signal generator widgets when calculating the DAPM context's target bias level. Otherwise the presence of a VMID widget, regardless whether it is on an active path or not, will cause the DAPM context to be powered up. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-dapm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c17c14c..177f8a1 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1840,6 +1840,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) */ switch (w->id) { case snd_soc_dapm_siggen: + case snd_soc_dapm_vmid: break; case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: -- 1.8.0

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[alsa-devel] [PATCH 0/2] ASoC: fix sound on mx31moboard
by Philippe Rétornaz 07 Oct '13

07 Oct '13

Hello Theses two patches are necessary to get back the sound working on mx31moboards. The first one fix imx-ssi to work even if only the dma is available. The second one fix the audmux settings on mx31moboard boards. Thanks, Philippe Philippe Rétornaz (2): ASoC: fsl: imx-ssi: fix probe on imx31 ASoC: fsl: Fix sound on mx31moboard sound/soc/fsl/imx-mc13783.c | 2 +- sound/soc/fsl/imx-ssi.c | 23 ++++++++++++----------- sound/soc/fsl/imx-ssi.h | 2 ++ 3 files changed, 15 insertions(+), 12 deletions(-) -- 1.7.9.5

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[alsa-devel] [PATCH 3/4] ESI W192M: Add sampling rate control.
by klem.dev＠gmail.com 07 Oct '13

07 Oct '13

From: Clement Guedez <klem.dev(a)gmail.com> Add sampling rate control for adc/dac in Ego Sys Waveterminal 192M. Signed-off-by: Clement Guedez <klem.dev(a)gmail.com> --- sound/pci/ice1712/wtm.c | 83 +++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 83 insertions(+) diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index d559916..cf2b6a5 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -29,6 +29,7 @@ #include <linux/interrupt.h> #include <linux/init.h> #include <sound/core.h> +#include <linux/slab.h> #include <sound/tlv.h> #include "ice1712.h" @@ -69,6 +70,37 @@ static inline unsigned char stac9460_2_get(struct snd_ice1712 *ice, int reg) /* * DAC mute control */ +static void stac9460_dac_mute_all(struct snd_ice1712 *ice, unsigned char mute, + unsigned char *change) +{ + unsigned char new, old; + int idx; + + /*stac9460 1*/ + for (idx = 0; idx < 7; idx++) { + if (change[idx]) { + old = stac9460_get(ice, idx); + new = (~mute << 7 & 0x80) | (old & ~0x80); + change[idx] = (new != old); + if (change[idx]) + stac9460_put(ice, idx, new); + } + } + + /*stac9460 2*/ + for (idx = 0; idx < 3; idx++) { + if (change[idx + 7]) { + old = stac9460_2_get(ice, idx); + new = (~mute << 7 & 0x80) | (old & ~0x80); + change[idx + 7] = (new != old); + if (change[idx + 7]) + stac9460_2_put(ice, idx, new); + } + } +} + + + #define stac9460_dac_mute_info snd_ctl_boolean_mono_info static int stac9460_dac_mute_get(struct snd_kcontrol *kcontrol, @@ -97,10 +129,13 @@ static int stac9460_dac_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct mutex *mute_mutex = ice->spec; unsigned char new, old; int id, idx; int change; + mutex_lock(mute_mutex); + if (kcontrol->private_value) { idx = STAC946X_MASTER_VOLUME; old = stac9460_get(ice, idx); @@ -128,6 +163,9 @@ static int stac9460_dac_mute_put(struct snd_kcontrol *kcontrol, stac9460_2_put(ice, idx - 6, new); } } + + mutex_unlock(mute_mutex); + return change; } @@ -381,6 +419,41 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, return change; } +/* + * Handler for setting correct codec rate - called when rate change is detected + */ +static void stac9460_set_rate_val(struct snd_ice1712 *ice, unsigned int rate) +{ + unsigned char old, new; + unsigned char changed[10]; + struct mutex *mute_mutex = ice->spec; + + if (rate == 0) /* no hint - S/PDIF input is master, simply return */ + return; + else if (rate <= 48000) + new = 0x08; /* 256x, base rate mode */ + else if (rate <= 96000) + new = 0x11; /* 256x, mid rate mode */ + else + new = 0x12; /* 128x, high rate mode */ + old = stac9460_get(ice, STAC946X_MASTER_CLOCKING); + if (old == new) + return; + /* change detected, setting master clock, muting first */ + /* due to possible conflicts with mute controls - mutexing */ + mutex_lock(mute_mutex); + /* we have to remember current mute status for each DAC */ + stac9460_dac_mute_all(ice, 0, changed); + /*printk(KERN_DEBUG "Rate change: %d, new MC: 0x%02x\n", rate, new);*/ + stac9460_put(ice, STAC946X_MASTER_CLOCKING, new); + udelay(10); + /* unmuting - only originally unmuted dacs - + * i.e. those changed when muting */ + stac9460_dac_mute_all(ice, 1, changed); + mutex_unlock(mute_mutex); +} + + static const DECLARE_TLV_DB_SCALE(db_scale_dac, -19125, 75, 0); static const DECLARE_TLV_DB_SCALE(db_scale_adc, 0, 150, 0); @@ -479,9 +552,11 @@ static int wtm_init(struct snd_ice1712 *ice) { static unsigned short stac_inits_wtm[] = { STAC946X_RESET, 0, + STAC946X_MASTER_CLOCKING, 0x11, (unsigned short)-1 }; unsigned short *p; + struct mutex *mute_mutex; /*WTM 192M*/ ice->num_total_dacs = 8; @@ -489,12 +564,20 @@ static int wtm_init(struct snd_ice1712 *ice) ice->force_rdma1 = 1; ice->vt1720 = 0; /* ice1724, e.g. 23 GPIOs */ + mute_mutex = kzalloc(sizeof(*mute_mutex), GFP_KERNEL); + if (!mute_mutex) + return -ENOMEM; + ice->spec = mute_mutex; + mutex_init(mute_mutex); + + /*initialize codec*/ p = stac_inits_wtm; for (; *p != (unsigned short)-1; p += 2) { stac9460_put(ice, p[0], p[1]); stac9460_2_put(ice, p[0], p[1]); } + ice->gpio.set_pro_rate = stac9460_set_rate_val; return 0; } -- 1.8.4.rc3

2 1

[alsa-devel] [PATCH 1/4] ESI W192M: Update ESI W192M driver.
by klem.dev＠gmail.com 07 Oct '13

07 Oct '13

From: Clement Guedez <klem.dev(a)gmail.com> Update Waveterminal 192M driver to support alsa changes that happened since last driver commit. Signed-off-by: Clement Guedez <klem.dev(a)gmail.com> --- sound/pci/ice1712/wtm.c | 37 ++++++++++++++++++++++--------------- 1 file changed, 22 insertions(+), 15 deletions(-) diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index bcf30a3..b6b650c 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -463,7 +463,7 @@ static int wtm_add_controls(struct snd_ice1712 *ice) static int wtm_init(struct snd_ice1712 *ice) { - static unsigned short stac_inits_prodigy[] = { + static unsigned short stac_inits_wtm[] = { STAC946X_RESET, 0, (unsigned short)-1 }; @@ -473,9 +473,10 @@ static int wtm_init(struct snd_ice1712 *ice) ice->num_total_dacs = 8; ice->num_total_adcs = 4; ice->force_rdma1 = 1; + ice->vt1720 = 0; /* ice1724, e.g. 23 GPIOs */ /*initialize codec*/ - p = stac_inits_prodigy; + p = stac_inits_wtm; for (; *p != (unsigned short)-1; p += 2) { stac9460_put(ice, p[0], p[1]); stac9460_2_put(ice, p[0], p[1]); @@ -485,19 +486,25 @@ static int wtm_init(struct snd_ice1712 *ice) static unsigned char wtm_eeprom[] = { - 0x47, /*SYSCONF: clock 192KHz, 4ADC, 8DAC */ - 0x80, /* ACLINK : I2S */ - 0xf8, /* I2S: vol; 96k, 24bit, 192k */ - 0xc1 /*SPDIF: out-en, spidf ext out*/, - 0x9f, /* GPIO_DIR */ - 0xff, /* GPIO_DIR1 */ - 0x7f, /* GPIO_DIR2 */ - 0x9f, /* GPIO_MASK */ - 0xff, /* GPIO_MASK1 */ - 0x7f, /* GPIO_MASK2 */ - 0x16, /* GPIO_STATE */ - 0x80, /* GPIO_STATE1 */ - 0x00, /* GPIO_STATE2 */ + [ICE_EEP2_SYSCONF] = 0x67, /* 49MHz crystal, mpu401, + * spdif-in+ 1 stereo ADC, + * 4 stereo DACs + */ + [ICE_EEP2_ACLINK] = 0x80, /* I2S */ + [ICE_EEP2_I2S] = 0xf8, /* vol, 96k, 24bit, 192k */ + [ICE_EEP2_SPDIF] = 0xc1, /* out-en, out-int, spdif-in */ + [ICE_EEP2_GPIO_DIR] = 0x9f, + [ICE_EEP2_GPIO_DIR1] = 0xff, + [ICE_EEP2_GPIO_DIR2] = 0x7f, + [ICE_EEP2_GPIO_MASK] = 0x9f, + [ICE_EEP2_GPIO_MASK1] = 0xff, + [ICE_EEP2_GPIO_MASK2] = 0x7f, + [ICE_EEP2_GPIO_STATE] = 0x16, + [ICE_EEP2_GPIO_STATE1] = 0x80, + [ICE_EEP2_GPIO_STATE2] = 0x00, /* GPIO20: 0 = CD drive dig. input + * passthrough, + * 1 = SPDIF-OUT from ice1724 + */ }; -- 1.8.4.rc3

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix mono speakers and headset mic on Dell Vostro 5470
by David Henningsson 07 Oct '13

07 Oct '13

On this machine, DAC on node 0x03 seems to give mono output. Also, it needs additional patches for headset mic support. It supports CTIA style headsets only. Alsa-info available at the bug link below. Cc: stable(a)kernel.org (v3.10+) BugLink: https://bugs.launchpad.net/bugs/1236228 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 27 +++++++++++++++++++++++++++ 1 file changed, 27 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0e303b9..52c26d3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3496,6 +3496,15 @@ static void alc282_fixup_asus_tx300(struct hda_codec *codec, } } +static void alc290_fixup_mono_speakers(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) + /* Remove DAC node 0x03, as it seems to be + giving mono output */ + snd_hda_override_wcaps(codec, 0x03, 0); +} + enum { ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, @@ -3522,6 +3531,7 @@ enum { ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT, ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, + ALC269_FIXUP_DELL3_MIC_NO_PRESENCE, ALC269_FIXUP_HEADSET_MODE, ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC, ALC269_FIXUP_ASUS_X101_FUNC, @@ -3535,6 +3545,7 @@ enum { ALC283_FIXUP_CHROME_BOOK, ALC282_FIXUP_ASUS_TX300, ALC283_FIXUP_INT_MIC, + ALC290_FIXUP_MONO_SPEAKERS, }; static const struct hda_fixup alc269_fixups[] = { @@ -3712,6 +3723,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC }, + [ALC269_FIXUP_DELL3_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1a, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC + }, [ALC269_FIXUP_HEADSET_MODE] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_mode, @@ -3804,6 +3824,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST }, + [ALC290_FIXUP_MONO_SPEAKERS] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc290_fixup_mono_speakers, + .chained = true, + .chain_id = ALC269_FIXUP_DELL3_MIC_NO_PRESENCE, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -3845,6 +3871,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0608, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0613, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0616, "Dell Vostro 5470", ALC290_FIXUP_MONO_SPEAKERS), SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- 1.7.9.5

2 1

[alsa-devel] [PATCH]: remove unnecessary/obsolete compat/hsearch_r.c
by Waldemar Brodkorb 07 Oct '13

07 Oct '13

The usage of hsearch functions where removed long time ago in commit 273d115de05574251bdd661747ecb68449a5cf1d. This patch highly increases portability for non-glibc systems. Remove the complete compat directory as requested by Takashi Iwai. Signed-off-by: Waldemar Brodkorb <wbx(a)openadk.org> --- configure.in | 4 +- src/Makefile.am | 7 +- src/compat/Makefile.am | 8 -- src/compat/hsearch_r.c | 236 ------------------------------------------------ 4 files changed, 3 insertions(+), 252 deletions(-) delete mode 100644 src/compat/Makefile.am delete mode 100644 src/compat/empty.c delete mode 100644 src/compat/hsearch_r.c diff --git a/configure.in b/configure.in index 00fff2b..35fd89f 100644 --- a/configure.in +++ b/configure.in @@ -64,8 +64,6 @@ AC_HEADER_TIME dnl Checks for library functions. AC_PROG_GCC_TRADITIONAL -AC_CHECK_FUNC([hsearch_r], [HAVE_HSEARCH_R=yes]) -AM_CONDITIONAL([ALSA_HSEARCH_R], [test "x$HAVE_HSEARCH_R" != xyes]) AC_CHECK_FUNCS([uselocale]) SAVE_LIBRARY_VERSION @@ -659,7 +657,7 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/pcm/Makefile src/pcm/scopes/Makefile \ src/rawmidi/Makefile src/timer/Makefile \ src/hwdep/Makefile src/seq/Makefile src/ucm/Makefile \ - src/compat/Makefile src/alisp/Makefile \ + src/alisp/Makefile \ src/conf/Makefile src/conf/alsa.conf.d/Makefile \ src/conf/cards/Makefile \ src/conf/pcm/Makefile \ diff --git a/src/Makefile.am b/src/Makefile.am index 08f482a..8f789fe 100644 --- a/src/Makefile.am +++ b/src/Makefile.am @@ -46,8 +46,8 @@ if BUILD_ALISP SUBDIRS += alisp libasound_la_LIBADD += alisp/libalisp.la endif -SUBDIRS += compat conf -libasound_la_LIBADD += compat/libcompat.la @ALSA_DEPLIBS@ +SUBDIRS += conf +libasound_la_LIBADD += @ALSA_DEPLIBS@ libasound_la_LDFLAGS = -version-info $(COMPATNUM) $(VSYMS) $(SYMFUNCS) $(LDFLAGS_NOUNDEFINED) @@ -84,9 +84,6 @@ ucm/libucm.la: instr/libinstr.la: $(MAKE) -C instr libinstr.la -compat/libcompat.la: - $(MAKE) -C compat libcompat.la - alisp/libalisp.la: $(MAKE) -C alisp libalisp.la diff --git a/src/compat/Makefile.am b/src/compat/Makefile.am deleted file mode 100644 index 01f54fc..0000000 --- a/src/compat/Makefile.am +++ /dev/null @@ -1,8 +0,0 @@ -noinst_LTLIBRARIES = libcompat.la -EXTRA_libcompat_la_SOURCES = hsearch_r.c - -if ALSA_HSEARCH_R -libcompat_la_SOURCES = empty.c hsearch_r.c -else -libcompat_la_SOURCES = empty.c -endif diff --git a/src/compat/empty.c b/src/compat/empty.c deleted file mode 100644 index e69de29..0000000 diff --git a/src/compat/hsearch_r.c b/src/compat/hsearch_r.c deleted file mode 100644 index 96ceac1..0000000 --- a/src/compat/hsearch_r.c +++ /dev/null @@ -1,236 +0,0 @@ -/* Copyright (C) 1993, 1995, 1996, 1997 Free Software Foundation, Inc. - This file is part of the GNU C Library. - Contributed by Ulrich Drepper <drepper(a)gnu.ai.mit.edu>, 1993. - - The GNU C Library is free software; you can redistribute it and/or - modify it under the terms of the GNU Lesser General Public License as - published by the Free Software Foundation; either version 2.1 of the - License, or (at your option) any later version. - - The GNU C Library is distributed in the hope that it will be useful, - but WITHOUT ANY WARRANTY; without even the implied warranty of - MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - Lesser General Public License for more details. - - You should have received a copy of the GNU Lesser General Public - License along with the GNU C Library; see the file COPYING.LIB. If not, - write to the Free Software Foundation, Inc., 59 Temple Place - Suite 330, - Boston, MA 02111-1307, USA. */ - -#include <errno.h> -#include <malloc.h> -#include <string.h> - -#define __USE_GNU -#ifndef __set_errno -#define __set_errno(e) errno = (e) -#endif -#include <search.h> - -/* [Aho,Sethi,Ullman] Compilers: Principles, Techniques and Tools, 1986 - [Knuth] The Art of Computer Programming, part 3 (6.4) */ - - -/* The reentrant version has no static variables to maintain the state. - Instead the interface of all functions is extended to take an argument - which describes the current status. */ -typedef struct _ENTRY -{ - unsigned int used; - ENTRY entry; -} -_ENTRY; - - -/* For the used double hash method the table size has to be a prime. To - correct the user given table size we need a prime test. This trivial - algorithm is adequate because - a) the code is (most probably) called a few times per program run and - b) the number is small because the table must fit in the core */ -static int -isprime (unsigned int number) -{ - /* no even number will be passed */ - unsigned int div = 3; - - while (div * div < number && number % div != 0) - div += 2; - - return number % div != 0; -} - - -/* Before using the hash table we must allocate memory for it. - Test for an existing table are done. We allocate one element - more as the found prime number says. This is done for more effective - indexing as explained in the comment for the hsearch function. - The contents of the table is zeroed, especially the field used - becomes zero. */ -int -hcreate_r (nel, htab) - size_t nel; - struct hsearch_data *htab; -{ - /* Test for correct arguments. */ - if (htab == NULL) - { - __set_errno (EINVAL); - return 0; - } - - /* There is still another table active. Return with error. */ - if (htab->table != NULL) - return 0; - - /* Change nel to the first prime number not smaller as nel. */ - nel |= 1; /* make odd */ - while (!isprime (nel)) - nel += 2; - - htab->size = nel; - htab->filled = 0; - - /* allocate memory and zero out */ - htab->table = (_ENTRY *) calloc (htab->size + 1, sizeof (_ENTRY)); - if (htab->table == NULL) - return 0; - - /* everything went alright */ - return 1; -} - - -/* After using the hash table it has to be destroyed. The used memory can - be freed and the local static variable can be marked as not used. */ -void -hdestroy_r (htab) - struct hsearch_data *htab; -{ - /* Test for correct arguments. */ - if (htab == NULL) - { - __set_errno (EINVAL); - return; - } - - if (htab->table != NULL) - /* free used memory */ - free (htab->table); - - /* the sign for an existing table is an value != NULL in htable */ - htab->table = NULL; -} - - -/* This is the search function. It uses double hashing with open addressing. - The argument item.key has to be a pointer to an zero terminated, most - probably strings of chars. The function for generating a number of the - strings is simple but fast. It can be replaced by a more complex function - like ajw (see [Aho,Sethi,Ullman]) if the needs are shown. - - We use an trick to speed up the lookup. The table is created by hcreate - with one more element available. This enables us to use the index zero - special. This index will never be used because we store the first hash - index in the field used where zero means not used. Every other value - means used. The used field can be used as a first fast comparison for - equality of the stored and the parameter value. This helps to prevent - unnecessary expensive calls of strcmp. */ -int -hsearch_r (item, action, retval, htab) - ENTRY item; - ACTION action; - ENTRY **retval; - struct hsearch_data *htab; -{ - unsigned int hval; - unsigned int count; - unsigned int len = strlen (item.key); - unsigned int idx; - - /* Compute an value for the given string. Perhaps use a better method. */ - hval = len; - count = len; - while (count-- > 0) - { - hval <<= 4; - hval += item.key[count]; - } - - /* First hash function: simply take the modulo but prevent zero. */ - hval %= htab->size; - if (hval == 0) - ++hval; - - /* The first index tried. */ - idx = hval; - - if (htab->table[idx].used) - { - /* Further action might be required according to the action value. */ - unsigned hval2; - - if (htab->table[idx].used == hval - && strcmp (item.key, htab->table[idx].entry.key) == 0) - { - if (action == ENTER) - htab->table[idx].entry.data = item.data; - - *retval = &htab->table[idx].entry; - return 1; - } - - /* Second hash function, as suggested in [Knuth] */ - hval2 = 1 + hval % (htab->size - 2); - - do - { - /* Because SIZE is prime this guarantees to step through all - available indexes. */ - if (idx <= hval2) - idx = htab->size + idx - hval2; - else - idx -= hval2; - - /* If we visited all entries leave the loop unsuccessfully. */ - if (idx == hval) - break; - - /* If entry is found use it. */ - if (htab->table[idx].used == hval - && strcmp (item.key, htab->table[idx].entry.key) == 0) - { - if (action == ENTER) - htab->table[idx].entry.data = item.data; - - *retval = &htab->table[idx].entry; - return 1; - } - } - while (htab->table[idx].used); - } - - /* An empty bucket has been found. */ - if (action == ENTER) - { - /* If table is full and another entry should be entered return - with error. */ - if (action == ENTER && htab->filled == htab->size) - { - __set_errno (ENOMEM); - *retval = NULL; - return 0; - } - - htab->table[idx].used = hval; - htab->table[idx].entry = item; - - ++htab->filled; - - *retval = &htab->table[idx].entry; - return 1; - } - - __set_errno (ESRCH); - *retval = NULL; - return 0; -} -- 1.7.9.5

2 1

[alsa-devel] ALSA lib pcm_hw.c:1466:(_snd_pcm_hw_open) Unknown field chmap
by Alexander Stein 07 Oct '13

07 Oct '13

Hi, I get this error message from several games I got recently. There is either not sound at all or it is choppy. This was with alsa-lib-1.0.27.1. So I downgraded to 1.0.26 and everything was fine. I compiled alsa-lib-1.0.27.1 myself and started them using LD_LIBRARY_PATH: everything worked. So I assumed there is something in the config files which has changed. Comparing them I noticed some changes in /usr/share/alsa/cards/Audigy2.conf. Those lines were added: chmap [ "UNKNOWN" "FL,FR" ] chmap [ "UNKNOWN" "RL,RR" ] chmap [ "UNKNOWN" "FC,LFE" ] chmap [ "UNKNOWN" "SL,SR" ] See alsa-lib commits 5ba11d69dafbd4502469fd957b4580169779664c and ec7acfc408856cfe71a122e3b5387463d7b3d919. Updating to 1.0.27.1 again and commenting those lines out everything is ok now. So I'm wondering what is the reason for this error message? How to fix that? Thanks, Alexander PS: my sound card: 03:07.0 Multimedia audio controller: Creative Labs SB0400 Audigy2 Value 1102:0008

2 1

[alsa-devel] [PATCH - Ego Sys Waveterminal 192M 1/1] Update driver to support alsa changes happened since last driver commit. Add TLV support for dB scales. Add sampling rate control.
by klem.dev＠gmail.com 05 Oct '13

05 Oct '13

From: Clement Guedez <klem.dev(a)gmail.com> Signed-off-by: Clement Guedez <klem.dev(a)gmail.com> diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index bcf30a3..286be47 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -4,8 +4,8 @@ * Lowlevel functions for Ego Sys Waveterminal 192M * * Copyright (c) 2006 Guedez Clement <klem.dev(a)gmail.com> - * Some functions are taken from the Prodigy192 driver - * source + * Thanks to the developers of the Prodigy192 driver on which + * I based this driver * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by @@ -25,15 +25,25 @@ +#include <linux/io.h> #include <linux/delay.h> #include <linux/interrupt.h> #include <linux/init.h> +#include <linux/slab.h> #include <sound/core.h> #include "ice1712.h" #include "envy24ht.h" #include "wtm.h" #include "stac946x.h" +#include <sound/tlv.h> + +struct wtm_spec { + struct ak4114 *ak4114; + /* rate change needs atomic mute/unmute of all dacs*/ + struct mutex mute_mutex; +}; + /* @@ -68,6 +78,35 @@ static inline unsigned char stac9460_2_get(struct snd_ice1712 *ice, int reg) /* * DAC mute control */ +static void stac9460_dac_mute_all(struct snd_ice1712 *ice, unsigned char mute, + unsigned char *change) +{ + unsigned char new, old; + int idx; + + /*stac9460 1*/ + for (idx = 0; idx < 7; idx++) { + if (change[idx]) { + old = stac9460_get(ice, idx); + new = (~mute << 7 & 0x80) | (old & ~0x80); + change[idx] = (new != old); + if (change[idx]) + stac9460_put(ice, idx, new); + } + } + + /*stac9460 2*/ + for (idx = 0; idx < 3; idx++) { + if (change[idx + 7]) { + old = stac9460_2_get(ice, idx); + new = (~mute << 7 & 0x80) | (old & ~0x80); + change[idx + 7] = (new != old); + if (change[idx + 7]) + stac9460_2_put(ice, idx, new); + } + } +} + #define stac9460_dac_mute_info snd_ctl_boolean_mono_info static int stac9460_dac_mute_get(struct snd_kcontrol *kcontrol, @@ -96,10 +135,14 @@ static int stac9460_dac_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct wtm_spec *spec = ice->spec; unsigned char new, old; int id, idx; int change; + mutex_lock(&spec->mute_mutex); + + if (kcontrol->private_value) { idx = STAC946X_MASTER_VOLUME; old = stac9460_get(ice, idx); @@ -127,6 +170,7 @@ static int stac9460_dac_mute_put(struct snd_kcontrol *kcontrol, stac9460_2_put(ice, idx - 6, new); } } + mutex_unlock(&spec->mute_mutex); return change; } @@ -175,7 +219,7 @@ static int stac9460_dac_vol_put(struct snd_kcontrol *kcontrol, if (kcontrol->private_value) { idx = STAC946X_MASTER_VOLUME; - nvol = ucontrol->value.integer.value[0] & 0x7f; + nvol = ucontrol->value.integer.value[0]; tmp = stac9460_get(ice, idx); ovol = 0x7f - (tmp & 0x7f); change = (ovol != nvol); @@ -186,7 +230,7 @@ static int stac9460_dac_vol_put(struct snd_kcontrol *kcontrol, } else { id = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); idx = id + STAC946X_LF_VOLUME; - nvol = ucontrol->value.integer.value[0] & 0x7f; + nvol = ucontrol->value.integer.value[0]; if (id < 6) tmp = stac9460_get(ice, idx); else @@ -339,7 +383,24 @@ static int stac9460_adc_vol_put(struct snd_kcontrol *kcontrol, * MIC / LINE switch fonction */ -#define stac9460_mic_sw_info snd_ctl_boolean_mono_info +static int stac9460_mic_sw_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static char *texts[2] = { "Line In", "Mic" }; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 2; + + if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) + uinfo->value.enumerated.item = uinfo->value.enumerated.items + - 1; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + + return 0; +} + static int stac9460_mic_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -353,7 +414,7 @@ static int stac9460_mic_sw_get(struct snd_kcontrol *kcontrol, val = stac9460_get(ice, STAC946X_GENERAL_PURPOSE); else val = stac9460_2_get(ice, STAC946X_GENERAL_PURPOSE); - ucontrol->value.integer.value[0] = ~val>>7 & 0x1; + ucontrol->value.enumerated.item[0] = (val >> 7) & 0x1; return 0; } @@ -369,7 +430,7 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, old = stac9460_get(ice, STAC946X_GENERAL_PURPOSE); else old = stac9460_2_get(ice, STAC946X_GENERAL_PURPOSE); - new = (~ucontrol->value.integer.value[0] << 7 & 0x80) | (old & ~0x80); + new = (ucontrol->value.enumerated.item[0] << 7 & 0x80) | (old & ~0x80); change = (new != old); if (change) { if (id == 0) @@ -380,6 +441,46 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, return change; } + +/* + * Handler for setting correct codec rate - called when rate change is detected + */ +static void stac9460_set_rate_val(struct snd_ice1712 *ice, unsigned int rate) +{ + unsigned char old, new; + unsigned char changed[10]; + struct wtm_spec *spec = ice->spec; + + if (rate == 0) /* no hint - S/PDIF input is master, simply return */ + return; + else if (rate <= 48000) + new = 0x08; /* 256x, base rate mode */ + else if (rate <= 96000) + new = 0x11; /* 256x, mid rate mode */ + else + new = 0x12; /* 128x, high rate mode */ + old = stac9460_get(ice, STAC946X_MASTER_CLOCKING); + if (old == new) + return; + /* change detected, setting master clock, muting first */ + /* due to possible conflicts with mute controls - mutexing */ + mutex_lock(&spec->mute_mutex); + /* we have to remember current mute status for each DAC */ + stac9460_dac_mute_all(ice, 0, changed); + /*printk(KERN_DEBUG "Rate change: %d, new MC: 0x%02x\n", rate, new);*/ + stac9460_put(ice, STAC946X_MASTER_CLOCKING, new); + udelay(10); + /* unmuting - only originally unmuted dacs - + * i.e. those changed when muting */ + stac9460_dac_mute_all(ice, 1, changed); + mutex_unlock(&spec->mute_mutex); +} + + + +static const DECLARE_TLV_DB_SCALE(db_scale_dac, -19125, 75, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_adc, 0, 150, 0); + /* * Control tabs */ @@ -390,15 +491,19 @@ static struct snd_kcontrol_new stac9640_controls[] = { .info = stac9460_dac_mute_info, .get = stac9460_dac_mute_get, .put = stac9460_dac_mute_put, - .private_value = 1 + .private_value = 1, + .tlv = { .p = db_scale_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Master Playback Volume", .info = stac9460_dac_vol_info, .get = stac9460_dac_vol_get, .put = stac9460_dac_vol_put, .private_value = 1, + .tlv = { .p = db_scale_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -419,11 +524,14 @@ static struct snd_kcontrol_new stac9640_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "DAC Volume", .count = 8, .info = stac9460_dac_vol_info, .get = stac9460_dac_vol_get, .put = stac9460_dac_vol_put, + .tlv = { .p = db_scale_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -435,18 +543,40 @@ static struct snd_kcontrol_new stac9640_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "ADC Volume", .count = 2, .info = stac9460_adc_vol_info, .get = stac9460_adc_vol_get, .put = stac9460_adc_vol_put, - + .tlv = { .p = db_scale_adc } } }; /*INIT*/ +static void stac9460_proc_regs_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct snd_ice1712 *ice = entry->private_data; + int reg, val; + /* registers 0x0 - 0x14 */ + for (reg = 0; reg <= 0x15; reg++) { + val = stac9460_get(ice, reg); + snd_iprintf(buffer, "0x%02x = 0x%02x\n", reg, val); + } +} + +static void stac9460_proc_init(struct snd_ice1712 *ice) +{ + struct snd_info_entry *entry; + if (!snd_card_proc_new(ice->card, "stac9460_codec", &entry)) + snd_info_set_text_ops(entry, ice, stac9460_proc_regs_read); +} + + static int wtm_add_controls(struct snd_ice1712 *ice) { unsigned int i; @@ -458,46 +588,64 @@ static int wtm_add_controls(struct snd_ice1712 *ice) if (err < 0) return err; } + stac9460_proc_init(ice); return 0; } static int wtm_init(struct snd_ice1712 *ice) { - static unsigned short stac_inits_prodigy[] = { + static unsigned short stac_inits_wtm[] = { STAC946X_RESET, 0, + STAC946X_MASTER_CLOCKING, 0x11, (unsigned short)-1 }; unsigned short *p; + struct wtm_spec *spec; + /*WTM 192M*/ ice->num_total_dacs = 8; ice->num_total_adcs = 4; ice->force_rdma1 = 1; + ice->vt1720 = 0; /* ice1724, e.g. 23 GPIOs */ + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + ice->spec = spec; + mutex_init(&spec->mute_mutex); /*initialize codec*/ - p = stac_inits_prodigy; + p = stac_inits_wtm; for (; *p != (unsigned short)-1; p += 2) { stac9460_put(ice, p[0], p[1]); stac9460_2_put(ice, p[0], p[1]); } + ice->gpio.set_pro_rate = stac9460_set_rate_val; return 0; } static unsigned char wtm_eeprom[] = { - 0x47, /*SYSCONF: clock 192KHz, 4ADC, 8DAC */ - 0x80, /* ACLINK : I2S */ - 0xf8, /* I2S: vol; 96k, 24bit, 192k */ - 0xc1 /*SPDIF: out-en, spidf ext out*/, - 0x9f, /* GPIO_DIR */ - 0xff, /* GPIO_DIR1 */ - 0x7f, /* GPIO_DIR2 */ - 0x9f, /* GPIO_MASK */ - 0xff, /* GPIO_MASK1 */ - 0x7f, /* GPIO_MASK2 */ - 0x16, /* GPIO_STATE */ - 0x80, /* GPIO_STATE1 */ - 0x00, /* GPIO_STATE2 */ + [ICE_EEP2_SYSCONF] = 0x67, /* 49MHz crystal, mpu401, + * spdif-in+ 1 stereo ADC, + * 4 stereo DACs + */ + [ICE_EEP2_ACLINK] = 0x80, /* I2S */ + [ICE_EEP2_I2S] = 0xf8, /* vol, 96k, 24bit, 192k */ + [ICE_EEP2_SPDIF] = 0xc1, /* out-en, out-int, spdif-in */ + [ICE_EEP2_GPIO_DIR] = 0x9f, + [ICE_EEP2_GPIO_DIR1] = 0xff, + [ICE_EEP2_GPIO_DIR2] = 0x7f, + [ICE_EEP2_GPIO_MASK] = 0x9f, + [ICE_EEP2_GPIO_MASK1] = 0xff, + [ICE_EEP2_GPIO_MASK2] = 0x7f, + [ICE_EEP2_GPIO_STATE] = 0x16, + [ICE_EEP2_GPIO_STATE1] = 0x80, + [ICE_EEP2_GPIO_STATE2] = 0x00, /* GPIO20: 0 = CD drive dig. input + * passthrough, + * 1 = SPDIF-OUT from ice1724 + */ }; -- 1.7.10.4

3 2

[alsa-devel] [PATCH 4/4] ESI W192M: Add text "Line in/Mic".
by klem.dev＠gmail.com 05 Oct '13

05 Oct '13

From: Clement Guedez <klem.dev(a)gmail.com> Add text "Line in/Mic" for selecting input gain state for Ego Sys Waveterminal 192M. Signed-off-by: Clement Guedez <klem.dev(a)gmail.com> --- sound/pci/ice1712/wtm.c | 21 ++++++++++++++++++--- 1 file changed, 18 insertions(+), 3 deletions(-) diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index cf2b6a5..464cca5 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -377,8 +377,23 @@ static int stac9460_adc_vol_put(struct snd_kcontrol *kcontrol, /* * MIC / LINE switch fonction */ +static int stac9460_mic_sw_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static char *texts[2] = { "Line In", "Mic" }; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 2; -#define stac9460_mic_sw_info snd_ctl_boolean_mono_info + if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) + uinfo->value.enumerated.item = uinfo->value.enumerated.items + - 1; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + + return 0; +} static int stac9460_mic_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -392,7 +407,7 @@ static int stac9460_mic_sw_get(struct snd_kcontrol *kcontrol, val = stac9460_get(ice, STAC946X_GENERAL_PURPOSE); else val = stac9460_2_get(ice, STAC946X_GENERAL_PURPOSE); - ucontrol->value.integer.value[0] = ~val>>7 & 0x1; + ucontrol->value.enumerated.item[0] = (val >> 7) & 0x1; return 0; } @@ -408,7 +423,7 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, old = stac9460_get(ice, STAC946X_GENERAL_PURPOSE); else old = stac9460_2_get(ice, STAC946X_GENERAL_PURPOSE); - new = (~ucontrol->value.integer.value[0] << 7 & 0x80) | (old & ~0x80); + new = (ucontrol->value.enumerated.item[0] << 7 & 0x80) | (old & ~0x80); change = (new != old); if (change) { if (id == 0) -- 1.8.4.rc3

1 0

[alsa-devel] [PATCH 2/4] ESI W192M: Add TLV Support.
by klem.dev＠gmail.com 05 Oct '13

05 Oct '13

From: Clement Guedez <klem.dev(a)gmail.com> Add TLV support for control value in dB scale for Ego Sys Waveterminal 192M. Signed-off-by: Clement Guedez <klem.dev(a)gmail.com> --- sound/pci/ice1712/wtm.c | 16 +++++++++++++++- 1 file changed, 15 insertions(+), 1 deletion(-) diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index b6b650c..d559916 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -29,6 +29,7 @@ #include <linux/interrupt.h> #include <linux/init.h> #include <sound/core.h> +#include <sound/tlv.h> #include "ice1712.h" #include "envy24ht.h" @@ -380,6 +381,10 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, return change; } + +static const DECLARE_TLV_DB_SCALE(db_scale_dac, -19125, 75, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_adc, 0, 150, 0); + /* * Control tabs */ @@ -394,11 +399,14 @@ static struct snd_kcontrol_new stac9640_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Master Playback Volume", .info = stac9460_dac_vol_info, .get = stac9460_dac_vol_get, .put = stac9460_dac_vol_put, .private_value = 1, + .tlv = { .p = db_scale_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -419,11 +427,14 @@ static struct snd_kcontrol_new stac9640_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "DAC Volume", .count = 8, .info = stac9460_dac_vol_info, .get = stac9460_dac_vol_get, .put = stac9460_dac_vol_put, + .tlv = { .p = db_scale_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -435,12 +446,15 @@ static struct snd_kcontrol_new stac9640_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "ADC Volume", .count = 2, .info = stac9460_adc_vol_info, .get = stac9460_adc_vol_get, .put = stac9460_adc_vol_put, - + .tlv = { .p = db_scale_adc } } }; -- 1.8.4.rc3

1 0

[alsa-devel] [PING!] Re: HDMI audio is not working on my new desktop
by Alexander E. Patrakov 05 Oct '13

05 Oct '13

2013/8/14 Alexander E. Patrakov <patrakov(a)gmail.com>: > Hello. > > I built a new desktop PC based on the Gigabyte H87N-WIFI motherboard. > It has one DVI and two HDMI outputs. However, I cannot get any sound > over HDMI. The same OS image works perfectly on my laptop that also > has an HDMI output. > > This is not a pulseaudio problem, as I don't get any sound even if I > test with this command: > > pasuspender -- speaker-test -c2 -D hdmi:0,2 # or, for that matter, 0,0 or 0,1 > > Alsa-info.sh has uploaded its output to > http://www.alsa-project.org/db/?f=1ad7192a15ff4a70b5ddb2579476dbb98adbd929 > > Please note this error in dmesg, it may be relevant: > > [ 1.803595] hda_codec: invalid CONNECT_LIST verb 5[1]:0 > [ 1.803719] hda_codec: invalid CONNECT_LIST verb 7[1]:0 As nobody replied, I have converted this into a formal bug report: https://bugzilla.kernel.org/show_bug.cgi?id=60769 Haswell people, _please_ comment on the bug. -- Alexander E. Patrakov

3 9

[alsa-devel] Microphone does not longer work on Lenovo N581
by Manuel Reimer 05 Oct '13

05 Oct '13

Hello, I'm using Archlinux and so usually run always on the latest kernel. But unfortunately somewhere with Kernel 3.9 my Intel sound got broken. Playback works, but no microphone is found. So I keep downgrading to 3.8.8 for some months, now. Can someone please help me with finding a solution for this? Is it really possible that the alsa driver "snd_had_intel" was broken between 3.8 and 3.9 and noone found this bug, so far? Thank you very much in advance Greetings, Manuel

3 4

[alsa-devel] [PATCH 1/6] ASoC: cs4271: Include linux/of.h header
by Sachin Kamat 04 Oct '13

04 Oct '13

'of_match_ptr' is defined in linux/of.h. Include it explicitly. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/codecs/cs4271.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index a20f1bb..f6e9534 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -25,6 +25,7 @@ #include <linux/gpio.h> #include <linux/i2c.h> #include <linux/spi/spi.h> +#include <linux/of.h> #include <linux/of_device.h> #include <linux/of_gpio.h> #include <sound/pcm.h> -- 1.7.9.5

2 9

[alsa-devel] us122l
by James Miller 04 Oct '13

04 Oct '13

please ignore my previous email -- James Stewart Miller Bsc(hons) Psych.

1 0

[alsa-devel] tascam us122l
by James Miller 04 Oct '13

04 Oct '13

Hi, I've just checked the kernel source for the tascam us122l (us122l.c us122l.h) and it has changed completely since the 3.4 kernel. I'm having a stack trace problem when trying to use jack with us122l. The card is recognised and I am using the .asoundrc from pub.briata.org as I always have done. When the us122l.c got broken (around about kernel version 3.6 to 3.9) I was able to get it working by recompiling the kernel after amending the us122l.c source code by changing a flag that was pushed in to a vm struct. In the new code I can't find any reference to the vm operation and it seems that the code has changed completely, I am going to try and revert to the 3.4 us122l.c to see if that will work, but I'd be grateful if someone could look into this. Thanks James -- James Stewart Miller Bsc(hons) Psych.

1 0

Re: [alsa-devel] [PATCH] sound: usb: support wireless devices in snd_usb_parse_datainterval
by Greg KH 04 Oct '13

04 Oct '13

On Tue, Oct 01, 2013 at 02:32:57PM -0500, Thomas Pugliese wrote: > Hi, > This patch adds support for dev speed USB_SPEED_WIRELESS in > snd_usb_parse_datainterval which allows the usb sound core to create > ISO urbs with the correct number and size of buffers. > > Signed-off-by: Thomas Pugliese <thomas.pugliese(a)gmail.com> Acked-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org>

1 0

Re: [alsa-devel] [PATCH] sound: usb: add support for wireless USB devices
by Greg KH 04 Oct '13

04 Oct '13

On Tue, Oct 01, 2013 at 02:32:15PM -0500, Thomas Pugliese wrote: > This patch updates snd_usb_audio_create also support devices whose > speed == USB_SPEED_WIRELESS. > > Signed-off-by: Thomas Pugliese <thomas.pugliese(a)gmail.com> Acked-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org>

1 0

[alsa-devel] snd_device_name_hint segfaults
by Forest Bond 03 Oct '13

03 Oct '13

Hi, I'm seeing segfaults with VLC similar to those reported here: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-January/059168.ht… This is on Ubuntu 12.04 with the latest alsa-lib package (1.0.25-1ubuntu10.2) and the following patches also applied: * 2cfc8b9b44a8e493c41b3d63d5a00b306a18a5ed snd_pcm_direct_parse_open_conf(): use thread-safe getgrnam_r() * 44c1a623dd1fc9e831616b663bebc54ca98df994 Add snd_lib_error_set_local() to install a thread-local error handler. * 25dbb102810b31c02358904d70d53c960fb0a10e snd_device_name_hint(): do not change the global error handler. * f49b2dc522a2564315c76d075203b15a39941e8a snd_device_name_hint(): do not use global snd_config. * 7f2b2c8c1650a1883b48abfcdb455138943854f9 conf: Fix a memory access violation resulting from improper error propogation So I think there are still issues when calling snd_device_name_hint(-1, ...). Here is the backtrace: #0 0x003f0e78 in __strcpy_chk () from /lib/i386-linux-gnu/libc.so.6 #1 0x0625084b in strcpy (__src=0x0, __dest=0xb6025e80 "pcm.(null)") at /usr/include/i386-li nux-gnu/bits/string3.h:105 #2 try_config (config=0xaf1c4df0, list=0xb52fe804, base=0x323e962 "pcm", name=0x0) at nameh int.c:243 #3 0x06251bc5 in add_software_devices (list=0xb52fe804, config=0xaf1c4df0) at namehint.c:51 2 #4 snd_device_name_hint (card=-1, iface=<optimized out>, hints=0xb52fe90c) at namehint.c:59 1 #5 0x0323c9bb in GetDevices (obj=0xb6271700, item=0x0, prefs_dev=<optimized out>) at alsa.c :721 #6 0x0323d4c0 in Probe (dev=0xb6011df8 "default", obj=0xb6271700) at alsa.c:161 #7 Open (obj=0xb6271700) at alsa.c:539 #8 0x0a6c73d0 in generic_start (func=0x323cd10, ap=0xb52fed98 "\273") at modules/modules.c:422 #9 0x0a6c7a3a in vlc_module_load (p_this=0xb6271700, psz_capability=0xa70b959 "audio output", psz_name=<optimized out>, b_strict=false, probe=0xa6c73c0 <generic_start>) at modules/modules.c:347 #10 0x0a6c8002 in module_need (obj=0xb6271700, cap=0xa70b959 "audio output", name=0xa70d2bd "$aout", strict=false) at modules/modules.c:437 #11 0x0a6b1242 in aout_OutputNew (p_aout=0xb6271700, p_format=0xb52feeac) at audio_output/output.c:59 #12 0x0a6ad77f in aout_DecNew (p_aout=0xb6271700, p_format=0xb52feeac, p_replay_gain=0xaef02a14, p_request_vout=0xb52feed0) at audio_output/dec.c:112 #13 0x0a67066e in aout_new_buffer (p_dec=0xaef02870, i_samples=1024) at input/decoder.c:2286 #14 0x0a672690 in decoder_NewAudioBuffer (p_decoder=0xaef02870, i_size=1024) at input/decoder.c:213 #15 0x04032f8c in DecodeBlock (p_dec=0xaef02870, pp_block=0xb52ff05c) at faad.c:464 #16 0x0a670891 in DecoderDecodeAudio (p_dec=0xaef02870, p_block=0xaef03000) at input/decoder.c:1291 #17 0x0a6716f5 in DecoderProcessAudio (b_flush=false, p_block=0xaef03000, p_dec=0xaef02870) at input/decoder.c:1936 #18 DecoderProcess (p_dec=0xaef02870, p_block=<optimized out>) at input/decoder.c:2059 #19 0x0a671959 in DecoderThread (p_data=0xaef02870) at input/decoder.c:938 #20 0x001fbd4c in start_thread () from /lib/i386-linux-gnu/libpthread.so.0 #21 0x003ddace in clone () from /lib/i386-linux-gnu/libc.so.6 As a work around I will probably patch VLC to not call snd_device_name_hint this way. I see this has been done for other applications: http://code.google.com/p/chromium/issues/detail?id=95797 http://code.mythtv.org/trac/ticket/10809 But I wanted to try diagnosing the issue first. Thoughts? Thanks, Forest -- Forest Bond http://www.alittletooquiet.net http://www.rapidrollout.com

4 6

[alsa-devel] [PATCH] ASoC: omap: Fix incorrect ARM dependency
by Jarkko Nikula 03 Oct '13

03 Oct '13

Commit b0e0a4d ("ASoC: omap: Enable COMPILE_TEST build for DT platforms") added two incorrect CONFIG_ARCH_ARM dependencies making impossible to select audio support for Nokia RX-51. Fix this by using correct CONFIG_ARM. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> --- This is a fix for 3.12-rc. I.e. not for next merge window. --- sound/soc/omap/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index daa78a0..4a07f71 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -1,6 +1,6 @@ config SND_OMAP_SOC tristate "SoC Audio for the Texas Instruments OMAP chips" - depends on (ARCH_OMAP && DMA_OMAP) || (ARCH_ARM && COMPILE_TEST) + depends on (ARCH_OMAP && DMA_OMAP) || (ARM && COMPILE_TEST) select SND_DMAENGINE_PCM config SND_OMAP_SOC_DMIC @@ -26,7 +26,7 @@ config SND_OMAP_SOC_N810 config SND_OMAP_SOC_RX51 tristate "SoC Audio support for Nokia RX-51" - depends on SND_OMAP_SOC && ARCH_ARM && (MACH_NOKIA_RX51 || COMPILE_TEST) + depends on SND_OMAP_SOC && ARM && (MACH_NOKIA_RX51 || COMPILE_TEST) select SND_OMAP_SOC_MCBSP select SND_SOC_TLV320AIC3X select SND_SOC_TPA6130A2 -- 1.8.4.rc3

2 1

[alsa-devel] [PATCH] ALSA: snd-usb-usx2y: fix start frame mask
by Daniel Mack 03 Oct '13

03 Oct '13

This patch is based on a change proposed by Guido Aulisi in a discussion on LAU which can be followed here: http://linuxaudio.org/mailarchive/lau/2013/5/20/200177 Cc: fzu(a)wemgehoertderstaat.de Original-code-by: Guido Aulisi <guido.aulisi(a)gmail.com> Tested-by: Dr Nicholas J Bailey <nicholas.bailey(a)glasgow.ac.uk> Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/usb/usx2y/usbusx2yaudio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 63fb521..cd16fcc 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -328,7 +328,7 @@ static void i_usX2Y_urb_complete(struct urb *urb) usX2Y_error_urb_status(usX2Y, subs, urb); return; } - if (likely((urb->start_frame & 0xFFFF) == (usX2Y->wait_iso_frame & 0xFFFF))) + if (likely((urb->start_frame & 0x03FF) == (usX2Y->wait_iso_frame & 0x03FF))) subs->completed_urb = urb; else { usX2Y_error_sequence(usX2Y, subs, urb); -- 1.8.3.1

3 4

[alsa-devel] [PATCH] ASoC: davinci-mcasp: add support for suspend and resume
by Daniel Mack 03 Oct '13

03 Oct '13

When the system returns from suspend, it looses its configuration. Most of it is restored by running a normal audio stream startup, but the DAI format is left unset as that's configured on the audio device creation. Hence, it suffices here to care for the registers which are touched by davinci_mcasp_set_dai_fmt() and restore them when the system is resumed. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/davinci/davinci-mcasp.c | 39 +++++++++++++++++++++++++++++++++++++++ sound/soc/davinci/davinci-mcasp.h | 12 ++++++++++++ 2 files changed, 51 insertions(+) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 591f853..5023da7 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1272,12 +1272,51 @@ static int davinci_mcasp_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM_SLEEP +static int davinci_mcasp_suspend(struct device *dev) +{ + struct davinci_audio_dev *a = dev_get_drvdata(dev); + void __iomem *base = a->base; + + a->context.txfmtctl = mcasp_get_reg(base + DAVINCI_MCASP_TXFMCTL_REG); + a->context.rxfmtctl = mcasp_get_reg(base + DAVINCI_MCASP_RXFMCTL_REG); + a->context.txfmt = mcasp_get_reg(base + DAVINCI_MCASP_TXFMT_REG); + a->context.rxfmt = mcasp_get_reg(base + DAVINCI_MCASP_RXFMT_REG); + a->context.aclkxctl = mcasp_get_reg(base + DAVINCI_MCASP_ACLKXCTL_REG); + a->context.aclkrctl = mcasp_get_reg(base + DAVINCI_MCASP_ACLKRCTL_REG); + a->context.pdir = mcasp_get_reg(base + DAVINCI_MCASP_PDIR_REG); + + return 0; +} + +static int davinci_mcasp_resume(struct device *dev) +{ + struct davinci_audio_dev *a = dev_get_drvdata(dev); + void __iomem *base = a->base; + + mcasp_set_reg(base + DAVINCI_MCASP_TXFMCTL_REG, a->context.txfmtctl); + mcasp_set_reg(base + DAVINCI_MCASP_RXFMCTL_REG, a->context.rxfmtctl); + mcasp_set_reg(base + DAVINCI_MCASP_TXFMT_REG, a->context.txfmt); + mcasp_set_reg(base + DAVINCI_MCASP_RXFMT_REG, a->context.rxfmt); + mcasp_set_reg(base + DAVINCI_MCASP_ACLKXCTL_REG, a->context.aclkxctl); + mcasp_set_reg(base + DAVINCI_MCASP_ACLKRCTL_REG, a->context.aclkrctl); + mcasp_set_reg(base + DAVINCI_MCASP_PDIR_REG, a->context.pdir); + + return 0; +} +#endif + +SIMPLE_DEV_PM_OPS(davinci_mcasp_pm_ops, + davinci_mcasp_suspend, + davinci_mcasp_resume); + static struct platform_driver davinci_mcasp_driver = { .probe = davinci_mcasp_probe, .remove = davinci_mcasp_remove, .driver = { .name = "davinci-mcasp", .owner = THIS_MODULE, + .pm = &davinci_mcasp_pm_ops, .of_match_table = mcasp_dt_ids, }, }; diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index a9ac0c1..a2e27e1 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -43,6 +43,18 @@ struct davinci_audio_dev { /* McASP FIFO related */ u8 txnumevt; u8 rxnumevt; + +#ifdef CONFIG_PM_SLEEP + struct { + u32 txfmtctl; + u32 rxfmtctl; + u32 txfmt; + u32 rxfmt; + u32 aclkxctl; + u32 aclkrctl; + u32 pdir; + } context; +#endif }; #endif /* DAVINCI_MCASP_H */ -- 1.8.3.1

3 2

[alsa-devel] [PATCH 1/3] ASoC: tas5086: move two variables into private struct
by Daniel Mack 03 Oct '13

03 Oct '13

We need to access the charge_period and start_mid_z values from other places later, so move them to the private struct. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/codecs/tas5086.c | 22 +++++++++++++--------- 1 file changed, 13 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index 6d31d88..31b5868 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -244,6 +244,8 @@ struct tas5086_private { unsigned int mclk, sclk; unsigned int format; bool deemph; + unsigned int charge_period; + unsigned int pwm_start_mid_z; /* Current sample rate for de-emphasis control */ int rate; /* GPIO driving Reset pin, if any */ @@ -720,13 +722,15 @@ static const int tas5086_charge_period[] = { static int tas5086_probe(struct snd_soc_codec *codec) { struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); - int charge_period = 1300000; /* hardware default is 1300 ms */ - u8 pwm_start_mid_z = 0; int i, ret; + priv->pwm_start_mid_z = 0; + priv->charge_period = 1300000; /* hardware default is 1300 ms */ + if (of_match_device(of_match_ptr(tas5086_dt_ids), codec->dev)) { struct device_node *of_node = codec->dev->of_node; - of_property_read_u32(of_node, "ti,charge-period", &charge_period); + of_property_read_u32(of_node, "ti,charge-period", + &priv->charge_period); for (i = 0; i < 6; i++) { char name[25]; @@ -735,7 +739,7 @@ static int tas5086_probe(struct snd_soc_codec *codec) "ti,mid-z-channel-%d", i + 1); if (of_get_property(of_node, name, NULL) != NULL) - pwm_start_mid_z |= 1 << i; + priv->pwm_start_mid_z |= 1 << i; } } @@ -744,25 +748,25 @@ static int tas5086_probe(struct snd_soc_codec *codec) * configure 'part 1' of the PWM starts to use Mid-Z, and tell * all configured mid-z channels to start start under 'part 1'. */ - if (pwm_start_mid_z) + if (priv->pwm_start_mid_z) regmap_write(priv->regmap, TAS5086_PWM_START, TAS5086_PWM_START_MIDZ_FOR_START_1 | - pwm_start_mid_z); + priv->pwm_start_mid_z); /* lookup and set split-capacitor charge period */ - if (charge_period == 0) { + if (priv->charge_period == 0) { regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, 0); } else { i = index_in_array(tas5086_charge_period, ARRAY_SIZE(tas5086_charge_period), - charge_period); + priv->charge_period); if (i >= 0) regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, i + 0x08); else dev_warn(codec->dev, "Invalid split-cap charge period of %d ns.\n", - charge_period); + priv->charge_period); } /* enable factory trim */ -- 1.8.3.1

2 3

[alsa-devel] [PATCH] ARM: omap: edma: add suspend suspend/resume hooks
by Daniel Mack 03 Oct '13

03 Oct '13

This patch adds support for edma driver. I know the intermediate code in arch/arm/common is going to be purged eventually when the real dma-engine driver lands, so I don't know whether it's reasonable to include my patch after all. It might just server as template for the suspend/resume functions in the final driver, or maybe anyone wants to include that intermediate hack for the time being. Successfully tested on an AM335x board with davinci-mcasp enabled audio. A patch to make the latter work after suspend has been posted on alsa-devel. Thanks, Dainel Daniel Mack (1): ARM: omap: edma: add suspend suspend/resume hooks arch/arm/common/edma.c | 133 +++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 133 insertions(+) -- 1.8.3.1

2 2

[alsa-devel] [PATCH] ALSA: snd-usb-usx2y: remove bogus frame check
by Daniel Mack 02 Oct '13

02 Oct '13

The frame check in i_usX2Y_urb_complete() is bogus and produces false positives as described in this LAU thread: http://linuxaudio.org/mailarchive/lau/2013/5/20/200177 This code removes the check code entirely. Cc: fzu(a)wemgehoertderstaat.de Reported-by: Dr Nicholas J Bailey <nicholas.bailey(a)glasgow.ac.uk> Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/usb/usx2y/usbusx2yaudio.c | 22 +++------------------- 1 file changed, 3 insertions(+), 19 deletions(-) diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 63fb521..6234a51 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -299,19 +299,6 @@ static void usX2Y_error_urb_status(struct usX2Ydev *usX2Y, usX2Y_clients_stop(usX2Y); } -static void usX2Y_error_sequence(struct usX2Ydev *usX2Y, - struct snd_usX2Y_substream *subs, struct urb *urb) -{ - snd_printk(KERN_ERR -"Sequence Error!(hcd_frame=%i ep=%i%s;wait=%i,frame=%i).\n" -"Most probably some urb of usb-frame %i is still missing.\n" -"Cause could be too long delays in usb-hcd interrupt handling.\n", - usb_get_current_frame_number(usX2Y->dev), - subs->endpoint, usb_pipein(urb->pipe) ? "in" : "out", - usX2Y->wait_iso_frame, urb->start_frame, usX2Y->wait_iso_frame); - usX2Y_clients_stop(usX2Y); -} - static void i_usX2Y_urb_complete(struct urb *urb) { struct snd_usX2Y_substream *subs = urb->context; @@ -328,12 +315,9 @@ static void i_usX2Y_urb_complete(struct urb *urb) usX2Y_error_urb_status(usX2Y, subs, urb); return; } - if (likely((urb->start_frame & 0xFFFF) == (usX2Y->wait_iso_frame & 0xFFFF))) - subs->completed_urb = urb; - else { - usX2Y_error_sequence(usX2Y, subs, urb); - return; - } + + subs->completed_urb = urb; + { struct snd_usX2Y_substream *capsubs = usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE], *playbacksubs = usX2Y->subs[SNDRV_PCM_STREAM_PLAYBACK]; -- 1.8.3.1

2 2

[alsa-devel] [PATCH] amixer: fix indentation when printing container TLV contents
by Anssi Hannula 02 Oct '13

02 Oct '13

decode_tlv() adds indentation in the beginning, with the expectation that the TLV will be printed on the line afterwards in a switch-case. However, in the case of a container TLV the switch-case simply adds another level of indentation and calls decode_tlv() for the inner TLVs. This causes the first inner TLV to be printed with too much indentation and double '|'. Fix that by printing "container" and a newline for container TLVs, so that the result is as follows: : values=0,0,0,0,0,0,0,0 | container | chmap-variable=FL,FR | chmap-variable=FL,FR,LFE | chmap-variable=FL,FR,FC Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> --- amixer/amixer.c | 1 + 1 file changed, 1 insertion(+) diff --git a/amixer/amixer.c b/amixer/amixer.c index b83b0c3..cf82892 100644 --- a/amixer/amixer.c +++ b/amixer/amixer.c @@ -460,6 +460,7 @@ static void decode_tlv(unsigned int spaces, unsigned int *tlv, unsigned int tlv_ } switch (type) { case SND_CTL_TLVT_CONTAINER: + printf("container\n"); size += sizeof(unsigned int) -1; size /= sizeof(unsigned int); while (idx < size) { -- 1.8.1.5

2 1

[alsa-devel] alsa-lib 1.0.27.2 and musl
by Waldemar Brodkorb 02 Oct '13

02 Oct '13

Hi Developers, I am trying to cross-compile alsa-lib with a musl¹ libc based toolchain. Unfortunately src/compat/hsearch_r.c is failing to compile, because of some glibc specific macros (like _P) normally defined in sys/cdefs.h. After grepping through the code I can not find any consumer for any functions implemented in hsearch_r.c. When I just comment out the code, everything compiles fine. Seems that that usage of hsearch functions was removed in the code long time ago. commit 273d115de05574251bdd661747ecb68449a5cf1d Author: Jaroslav Kysela <perex(a)perex.cz> Date: Sat Jul 15 10:20:32 2000 +0000 Removed old mixer API functions. Added simple control layer (experimental). Can it just be removed? thanks for any advice or comment. best regards Waldemar ¹ http://www.musl-libc.org/

3 2

Re: [alsa-devel] [PATCH] sound: usb: add support for wireless USB devices
by Daniel Mack 02 Oct '13

02 Oct '13

On 01.10.2013 21:32, Thomas Pugliese wrote: > This patch updates snd_usb_audio_create also support devices whose > speed == USB_SPEED_WIRELESS. Interesting. What device are you testing this with? Daniel > > Signed-off-by: Thomas Pugliese <thomas.pugliese(a)gmail.com> > --- > sound/usb/card.c | 1 + > 1 file changed, 1 insertion(+) > > diff --git a/sound/usb/card.c b/sound/usb/card.c > index 64952e2..ef76aa0 100644 > --- a/sound/usb/card.c > +++ b/sound/usb/card.c > @@ -349,6 +349,7 @@ static int snd_usb_audio_create(struct usb_device *dev, int idx, > case USB_SPEED_LOW: > case USB_SPEED_FULL: > case USB_SPEED_HIGH: > + case USB_SPEED_WIRELESS: > case USB_SPEED_SUPER: > break; > default: >

2 1

[alsa-devel] snd-usb-usx2y (Tascam US-122) sequence errors
by Dr Nicholas J Bailey 02 Oct '13

02 Oct '13

I'm on debian jessie, Linux arial 3.10-3-686-pae #1 SMP Debian 3.10.11-1 (2013-09-10) i686 GNU/Linux with a Tascam us-122 that has recently stopped working. When I try to capture with anything, there is a few hundred ms of audio then the capture process stalls and the following dmesg appears: [ 462.596296] Sequence Error!(hcd_frame=0 ep=8in;wait=1024,frame=0). [ 462.596296] Most probably some urb of usb-frame 1024 is still missing. [ 462.596296] Cause could be too long delays in usb-hcd interrupt handling. [ 477.959230] Sequence Error!(hcd_frame=0 ep=8in;wait=1024,frame=0). [ 477.959230] Most probably some urb of usb-frame 1024 is still missing. [ 477.959230] Cause could be too long delays in usb-hcd interrupt handling. Before this kernel, the box worked OK. The only think that leads me to believe it might be my fault is that for a very long time, I've had to plug the box in: [ 445.529271] usb 2-1.6: new full-speed USB device number 5 using ehci-pci [ 445.622076] usb 2-1.6: New USB device found, idVendor=1604, idProduct=8006 [ 445.622086] usb 2-1.6: New USB device strings: Mfr=0, Product=0, SerialNumber=0 then unplug it and plug it in again: [ 445.972743] usb 2-1.6: USB disconnect, device number 5 [ 447.621519] usb 2-1.6: new full-speed USB device number 6 using ehci-pci [ 447.714462] usb 2-1.6: New USB device found, idVendor=1604, idProduct=8007 [ 447.714471] usb 2-1.6: New USB device strings: Mfr=0, Product=0, SerialNumber=0 [ 447.834852] usbcore: registered new interface driver snd-usb-usx2y [ 455.194314] usb 2-1.6: USB disconnect, device number 6 [ 457.487433] usb 2-1.6: new full-speed USB device number 7 using ehci-pci [ 457.579960] usb 2-1.6: New USB device found, idVendor=1604, idProduct=8006 [ 457.579969] usb 2-1.6: New USB device strings: Mfr=0, Product=0, SerialNumber=0 to get the firmware to load [ 457.888884] usb 2-1.6: USB disconnect, device number 7 [ 459.379782] usb 2-1.6: new full-speed USB device number 8 using ehci-pci [ 459.472516] usb 2-1.6: New USB device found, idVendor=1604, idProduct=8007 [ 459.472526] usb 2-1.6: New USB device strings: Mfr=0, Product=0, SerialNumber=0 (green USB light comes on here). I'm not sure about how to start debugging this; there are many things that could be wrong. If it will help, I'll try and file a bug report; if not please tell me to shut up (prefereably after pointing me at the solution :) ) Nick/.

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Re: [alsa-devel] Realtek ALC286 SONY Vaio Pro 13 Ubuntu 13.10
by David Henningsson 02 Oct '13

02 Oct '13

On 10/02/2013 04:13 AM, Yuval Levy wrote: > On 10/01/2013 02:28 AM, David Henningsson wrote: >> Ok. If things work without extra drivers from Sony or Realtek, then it >> should be possible to get things working in Linux too without extra >> information from these parties... > > I have added this information to the bug report > <https://bugs.launchpad.net/bugs/1228689> > where I also post my comments so that they are more persistent and > available to other users with similar issues. > > >> Ok - given that 0x21 works as expected you should only need to do this >> for pin 0x18. > > All my testing was with doing this for both pins. I only noticed now > that I do not need to do this for pin 0x18. I will try to o this only > for pin 0x21 and reboot tomorrow. Ok, cool. Notice that I wrote "do this for pin 0x18 only, not for pin 0x21", where as you replied "I'll do this for pin 0x21 only, not for pin 0x18". 0x18 is for input switching, 0x21 is for output switching. If you remove jack detection, like you did, you will need to do this switching manually, e g in pavucontrol or gnome sound settings. That is, when you plug your headphone in, you need to manually select "Headphone" too, if you don't, the result will probably be that you hear sound in both headphones and speakers. Anyway, let's guess that the high pitched sound, as well as the lack of headset microphone input, is because your headset is OMTP and that the Vaio only supports CTIA. That's my *guess*, but it could be wrong, and it would be good to have this confirmed with a CTIA headset. -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

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[alsa-devel] Problems with i.MX233 driver in 3.12rc2
by Trevor Page 01 Oct '13

01 Oct '13

Hello, First post to this list - I hope this is the appropriate place to ask for this assistance. I'm using the Olinuxino Maxi board with the i.MX233 processor. I'm using the 3.12rc2 kernel at the moment because this has recently introduced the drivers for this processor's audio. I understand that the drivers for this chip's audio are: sound/soc/codecs/mxs-builtin-codec.c sound/soc/mxs/mxs-builtin-audio.c sound/soc/mxs/mxs-builtin-dai.c sound/soc/mxs/mxs-builtin-pcm.c My application for the board is a simple C program that uses the Linphone library to make VOIP calls. The problem I am having is that audio usually works (playback and capture) perfectly fine from boot, but at some point it may suddenly go 'dead'. At this point, there is no longer any audio from the headphone socket. I have not yet pinpointed exactly what it takes to trigger the fault, but I have seen it happen once when the call was put on hold while a Linphone call was in progress. Once the audio output is dead, these are some further observations: - When a Linphone call is received, I still hear a click from the headphone output when the call ends. - I have confirmed that the DAC / HP channels are turned up and unmuted. - If I run speaker-test when the audio is working, speaker-test periodically prints time duration information (presumably on each pass of its internal loop), like "0 - Front Left", "Time per period ...". When the audio is dead, speaker-test just freezes on "0 - Front Left" forever, as if some call is either blocking or returning to say it is busy. This might hopefully therefore be a useful point to start debugging. - I have only ever managed to restore audio using a reboot. I would like to have a go at debugging this. My problem though is that although I'm experienced with C and microprocessor programming, compiling and debugging embedded Linux is new ground for me. I'd be grateful for any pointers on things I could try in order to gain an insight into what's going wrong, and tips on how I could debug the drivers. So far my plan for today is to build the kernel again with the i.MX233 audio drivers as external modules so that I can hopefully play with them a bit and compile them separately. (I assume the development and debug process for audio drivers like this is to compile them as .ko modules and modprobe them in). However without having any prior knowledge of Linux drivers or the ALSA / ASoC framework I'm not sure if I'm going to get very far. Regards, Trevor.

3 2

[alsa-devel] [PATCH] ASoC: kirkwood: fix compilation warning in kirkwood_dma_open
by Vladimir Murzin 01 Oct '13

01 Oct '13

writel() supposes the first argument of type unsigned int. This fix the warning: sound/soc/kirkwood/kirkwood-dma.c: In function 'kirkwood_dma_open': sound/soc/kirkwood/kirkwood-dma.c:164:3: warning: large integer implicitly truncated to unsigned type [-Woverflow] Signed-off-by: Vladimir Murzin <murzin.v(a)gmail.com> --- sound/soc/kirkwood/kirkwood-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index b238434..1135c30 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -161,7 +161,7 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream) * Enable Error interrupts. We're only ack'ing them but * it's useful for diagnostics */ - writel((unsigned long)-1, priv->io + KIRKWOOD_ERR_MASK); + writel((unsigned int)-1, priv->io + KIRKWOOD_ERR_MASK); } dram = mv_mbus_dram_info(); -- 1.8.1.5

2 1

[alsa-devel] [PATCH] ASoC: fsl: imx-sgtl5000: Add .remove back
by Fabio Estevam 01 Oct '13

01 Oct '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Commit e8f00c1b01 (Merge remote-tracking branch 'asoc/fix/fsl' into asoc-devm) fixed a conflict, but missed to add the .remove function back,which causes the following build warning: sound/soc/fsl/imx-sgtl5000.c:185:12: warning: 'imx_sgtl5000_remove' defined but not used [-Wunused-function] Fix the warning by adding the .remove function back. Reported-by: Olof Johansson <olof(a)lixom.net> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/imx-sgtl5000.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 6f4bdc8..ed6ba1e 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -204,6 +204,7 @@ static struct platform_driver imx_sgtl5000_driver = { .of_match_table = imx_sgtl5000_dt_ids, }, .probe = imx_sgtl5000_probe, + .remove = imx_sgtl5000_remove, }; module_platform_driver(imx_sgtl5000_driver); -- 1.8.1.2

2 1

[alsa-devel] [PATCH] snd_tlv_convert_from_dB: fix decreasing gain across entries
by Stephen Warren 01 Oct '13

01 Oct '13

From: Stephen Warren <swarren(a)nvidia.com> Currently, for a TLV consisting of TLV_DB_SCALE_ITEMs, if e.g. alsamixer calls snd_mixer_selem_set_playback_dB() with a value that is in-between two TLV_DB_SCALE_ITEMs, and xdir is negative, the selected raw hardware value is the minimum in the first range above that value, rather than the maximum in the last range below that value. The user-visible symptom is that in alsamixer, pressing the down key to reduce the value sticks at certain points, and cannot be incrementally reduced any further, although directly selecting a much lower value (e.g. by pressing 0..9) works as expected. This is triggered e.g. by sound/soc/codec/max98090.c's max98090_hp_tlv[]. Fix this by checking whether xdir is positive or not, rather than checking whether it has a non-zero value. The code to select the previous range's max value is already present. This matches how xdir is used in other parts of the code. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- src/control/tlv.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/control/tlv.c b/src/control/tlv.c index 6b0b9f4..b08d887 100644 --- a/src/control/tlv.c +++ b/src/control/tlv.c @@ -312,7 +312,7 @@ int snd_tlv_convert_from_dB(unsigned int *tlv, long rangemin, long rangemax, submin, submax, db_gain, value, xdir); else if (db_gain < dbmin) { - *value = xdir || pos == 2 ? submin : prev_submax; + *value = xdir > 0 || pos == 2 ? submin : prev_submax; return 0; } prev_submax = submax; -- 1.8.1.5

2 1

[alsa-devel] [PATCH] amixer: actually print all TLVs in a container TLV
by Anssi Hannula 01 Oct '13

01 Oct '13

Due to a wrong indexing of tlv[] decode_tlv() always skips to the end of the container TLV since the used tlv[1] contains the container TLV size instead of the intended element size. This causes, for example, only the first HDMI channel map TLV to be shown. Fix the indexing to actually use the element size. Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> --- amixer/amixer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/amixer/amixer.c b/amixer/amixer.c index 7727943..b83b0c3 100644 --- a/amixer/amixer.c +++ b/amixer/amixer.c @@ -468,7 +468,7 @@ static void decode_tlv(unsigned int spaces, unsigned int *tlv, unsigned int tlv_ return; } decode_tlv(spaces + 2, tlv + idx, tlv[idx+1] + 8); - idx += 2 + (tlv[1] + sizeof(unsigned int) - 1) / sizeof(unsigned int); + idx += 2 + (tlv[idx+1] + sizeof(unsigned int) - 1) / sizeof(unsigned int); } break; case SND_CTL_TLVT_DB_SCALE: -- 1.8.1.5

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Re: [alsa-devel] Bug: Conexant CX20588, Kernel 3.11, Win8 changes GPIO settings
by Takashi Iwai 01 Oct '13

01 Oct '13

At Mon, 30 Sep 2013 21:00:38 +0200, Christopher K. wrote: > > Works fine. Thanks for your support. I will now dive into the code to > be able to send you a patch the next time something is wrong :) OK, applied it now. thanks, Takashi > > > Chris > > On Mon, Sep 30, 2013 at 12:19 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Fri, 27 Sep 2013 22:38:22 +0200, > > Christopher wrote: > >> > >> I attached the alsa-info. This issue seems to be specific to the driver > >> provided by Acer (with thx, equalizer etc.) > >> Thanks for your help. > > > > Thanks. Could you check whether the patch below works? > > > > > > Takashi > > > > --- > > From: Takashi Iwai <tiwai(a)suse.de> > > Subject: [PATCH] ALSA: hda - Fix GPIO for Acer Aspire 3830TG > > > > Acer Aspire 3830TG seems requiring GPIO bit 0 as the primary mute > > control. When a machine is booted after Windows 8, the GPIO pin is > > turned off and it results in the silent output. > > > > This patch adds the manual fixup of GPIO bit 0 for this model. > > > > Reported-by: Christopher <DIDI2002(a)web.de> > > Cc: <stable(a)vger.kernel.org> > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > > --- > > sound/pci/hda/patch_conexant.c | 11 +++++++++++ > > 1 file changed, 11 insertions(+) > > > > diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c > > index 4edd2d0..ec68eae 100644 > > --- a/sound/pci/hda/patch_conexant.c > > +++ b/sound/pci/hda/patch_conexant.c > > @@ -3231,6 +3231,7 @@ enum { > > CXT_FIXUP_INC_MIC_BOOST, > > CXT_FIXUP_HEADPHONE_MIC_PIN, > > CXT_FIXUP_HEADPHONE_MIC, > > + CXT_FIXUP_GPIO1, > > }; > > > > static void cxt_fixup_stereo_dmic(struct hda_codec *codec, > > @@ -3375,6 +3376,15 @@ static const struct hda_fixup cxt_fixups[] = { > > .type = HDA_FIXUP_FUNC, > > .v.func = cxt_fixup_headphone_mic, > > }, > > + [CXT_FIXUP_GPIO1] = { > > + .type = HDA_FIXUP_VERBS, > > + .v.verbs = (const struct hda_verb[]) { > > + { 0x01, AC_VERB_SET_GPIO_MASK, 0x01 }, > > + { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01 }, > > + { 0x01, AC_VERB_SET_GPIO_DATA, 0x01 }, > > + { } > > + }, > > + }, > > }; > > > > static const struct snd_pci_quirk cxt5051_fixups[] = { > > @@ -3384,6 +3394,7 @@ static const struct snd_pci_quirk cxt5051_fixups[] = { > > > > static const struct snd_pci_quirk cxt5066_fixups[] = { > > SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC), > > + SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT_FIXUP_GPIO1), > > SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), > > SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), > > SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T410", CXT_PINCFG_LENOVO_TP410), > > -- > > 1.8.4 > > >

1 0

Re: [alsa-devel] Realtek ALC286 SONY Vaio Pro 13 Ubuntu 13.10
by David Henningsson 01 Oct '13

01 Oct '13

On 10/01/2013 08:13 AM, Yuval Levy wrote: > Thank you for the quick reply. > > On 09/30/2013 03:50 AM, David Henningsson wrote: >> An important distinction is here: to they work in Windows without >> specific Realtek drivers installed, or do you require Realtek's Windows >> drivers for this support? > > I am 99% certain that yes. I am no Windows expert. After taking > delivery of the laptop and ascertaining that the functionality that I > need works, I have backed up SONY's original stuff and stored it away. > Far away. I have reformatted the SDD, leaving 30 GB for a Windows > install (because unfortunately, the law school's exam system requires > either Windows or Mac). I reinstalled Windows from Microsoft's media. > I have not installed any other driver myself, but I am not sure if > Windows downloads drivers from the internet. It definitely did download > drivers for my Brother printer, automatically and without asking for > permission, after it discovered it on the network. I booted into > Windows before writing this email and looked in the Control Panel. It > says the driver for the two microsoft and for the loudspeakers is Microsoft. Ok. If things work without extra drivers from Sony or Realtek, then it should be possible to get things working in Linux too without extra information from these parties... >> Second, to do some actual triaging. I see you have been running >> hda-jack-sense-test. Could you run it all three scenarios: >> 1) nothing connected >> 2) headphone (without mic) connected >> 3) headset (heaphone + mic combo) connected > > Done and added to the bug report. 0x21 works as you expected. 0x18 not. > > As you suggested I installed alsa-tools-gui and did the advance override > with hdajackretask: > * selected codec Realtek ALC286 > * clicked Advance override in the Options on the right hand side > * clicked Override under pins 0x18 and 0x21 > * changed Jack detection from Present to Not present for both pins Ok - given that 0x21 works as expected you should only need to do this for pin 0x18. > * clicked Apply now > > An error message appeared: Failed to create file > '/home/yuv/.pulse/client.conf.3K083W': No such file or directory > > maybe it should save to /home/yuv/.config/pulse/client.conf.3K083W? > ^^^^^^^ > I am sorry I do not know how to fix this, can you help? Oh. That's fixed upstream [1], but apparently not in Ubuntu yet. Any of these workarounds should do it: 1) use the "boot override" instead (and reboot your computer to test) 2) create a symlink from .pulse to .config/pulse 3) or copy everything from .config/pulse to .pulse (but then you have to restart your computer afterwards, so PulseAudio will look in .pulse instead). -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic [1] http://git.alsa-project.org/?p=alsa-tools.git;a=commitdiff;h=772fbde91c00c8…

1 0

[alsa-devel] [PATCH v2] ASoC: imx-sgtl5000: do not use devres on a foreign device
by Philipp Zabel 01 Oct '13

01 Oct '13

Calling devm_clk_get with any device pointer other than our own confuses devres. Use clk_get instead. This avoids hitting the following warning in the imx-sgtl5000 error path: imx-sgtl5000 sound.12: snd_soc_register_card failed (-517) platform sound.12: Driver imx-sgtl5000 requests probe deferral ------------[ cut here ]------------ WARNING: CPU: 0 PID: 75 at drivers/base/dd.c:272 driver_probe_device+0x194/0x218() Modules linked in: snd_soc_sgtl5000(+) snd_soc_imx_sgtl5000 coda snd_soc_imx_audmux imx_sdma snd_soc_fsl_spdif snd_soc_fsl_ssi CPU: 0 PID: 75 Comm: udevd Not tainted 3.11.0-rc6+ #4682 Backtrace: [<80010bc4>] (dump_backtrace+0x0/0x10c) from [<80010d60>] (show_stack+0x18/0x1c) r6:00000110 r5:00000009 r4:00000000 r3:00000000 [<80010d48>] (show_stack+0x0/0x1c) from [<804f0764>] (dump_stack+0x20/0x28) [<804f0744>] (dump_stack+0x0/0x28) from [<8001a4a4>] (warn_slowpath_common+0x6c/0x8c) [<8001a438>] (warn_slowpath_common+0x0/0x8c) from [<8001a4e8>] (warn_slowpath_null+0x24/0x2c) r8:7f032000 r7:7f02f93c r6:cf8eaa54 r5:cf8eaa20 r4:80728a0c [<8001a4c4>] (warn_slowpath_null+0x0/0x2c) from [<80286bdc>] (driver_probe_device+0x194/0x218) [<80286a48>] (driver_probe_device+0x0/0x218) from [<80286cf4>] (__driver_attach+0x94/0x98) r7:00000000 r6:cf8eaa54 r5:7f02f93c r4:cf8eaa20 [<80286c60>] (__driver_attach+0x0/0x98) from [<802851c8>] (bus_for_each_dev+0x5c/0x90) r6:80286c60 r5:7f02f93c r4:00000000 r3:cf8ef03c [<8028516c>] (bus_for_each_dev+0x0/0x90) from [<80286654>] (driver_attach+0x24/0x28) r6:806d0424 r5:cf16a580 r4:7f02f93c [<80286630>] (driver_attach+0x0/0x28) from [<802861e4>] (bus_add_driver+0xdc/0x234) [<80286108>] (bus_add_driver+0x0/0x234) from [<802871d4>] (driver_register+0x80/0x154) r8:7f032000 r7:00000001 r6:7f02fa68 r5:7f02fa74 r4:7f02f93c [<80287154>] (driver_register+0x0/0x154) from [<8033c278>] (i2c_register_driver+0x34/0xbc) [<8033c244>] (i2c_register_driver+0x0/0xbc) from [<7f032018>] (sgtl5000_i2c_driver_init+0x18/0x24 [snd_soc_sgtl5000]) r5:7f02fa74 r4:cfb7ff48 [<7f032000>] (sgtl5000_i2c_driver_init+0x0/0x24 [snd_soc_sgtl5000]) from [<80008738>] (do_one_initcall+0xf4/0x150) [<80008644>] (do_one_initcall+0x0/0x150) from [<80053f64>] (load_module+0x174c/0x1db4) [<80052818>] (load_module+0x0/0x1db4) from [<800546ac>] (SyS_init_module+0xe0/0xf4) [<800545cc>] (SyS_init_module+0x0/0xf4) from [<8000e540>] (ret_fast_syscall+0x0/0x30) r6:00005b22 r5:00afed68 r4:00000000 ---[ end trace b24c5c3bb145dbdd ]--- Signed-off-by: Philipp Zabel <p.zabel(a)pengutronix.de> Acked-by: Shawn Guo <shawn.guo(a)linaro.org> Reviewed-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Rebased onto asoc/for-next, adding back imx_sgtl5000_remove. --- sound/soc/fsl/imx-sgtl5000.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 78f86d8..c49e4cb 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -128,7 +128,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) goto fail; } - data->codec_clk = devm_clk_get(&codec_dev->dev, NULL); + data->codec_clk = clk_get(&codec_dev->dev, NULL); if (IS_ERR(data->codec_clk)) { ret = PTR_ERR(data->codec_clk); goto fail; @@ -172,6 +172,8 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) return 0; fail: + if (!IS_ERR(data->codec_clk)) + clk_put(data->codec_clk); if (ssi_np) of_node_put(ssi_np); if (codec_np) @@ -180,6 +182,15 @@ fail: return ret; } +static int imx_sgtl5000_remove(struct platform_device *pdev) +{ + struct imx_sgtl5000_data *data = platform_get_drvdata(pdev); + + clk_put(data->codec_clk); + + return 0; +} + static const struct of_device_id imx_sgtl5000_dt_ids[] = { { .compatible = "fsl,imx-audio-sgtl5000", }, { /* sentinel */ } -- 1.8.4.rc3

3 2

[alsa-devel] [PATCH 2/2] [RESEND] snd-ac97-codec: Use complete ID when the codec is not known
by Ondrej Zary 30 Sep '13

30 Sep '13

Currently, if the codec ID is not known, the last byte of the ID is used as name. It's not helpful at all, use full codec ID instead. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- sound/pci/ac97/ac97_codec.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index bf578ba2..30a40a8 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -1830,7 +1830,7 @@ void snd_ac97_get_name(struct snd_ac97 *ac97, unsigned int id, char *name, int m pid->patch(ac97); } } else - sprintf(name + strlen(name), " id %x", id & 0xff); + sprintf(name + strlen(name), " id %x", id); } /** -- Ondrej Zary

2 2

[alsa-devel] [PATCH] ASoC: generic: Add generic DT based simple codec
by Jean-Francois Moine 30 Sep '13

30 Sep '13

This patch adds a simple sound codec which is described by the DT. This codec may be used when no specific codec action is needed. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- .../devicetree/bindings/sound/simple-codec.txt | 103 +++++++++++++ sound/soc/generic/Kconfig | 6 + sound/soc/generic/Makefile | 2 + sound/soc/generic/simple-codec.c | 197 +++++++++++++++++++++++++ 4 files changed, 3088 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/simple-codec.txt b/Documentation/devicetree/bindings/sound/simple-codec.txt new file mode 100644 index 0000000..75be747 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/simple-codec.txt @@ -0,0 +1,103 @@ +Device-Tree bindings for the simple codec + +Required properties: +- compatible: should be "linux,simple-codec". +- dai-name: name of the codec + +Optional properties: +- capture: information about capture +- playback: information about playback +At least one of the 'capture' or 'playback' nodes must be present. + +Child 'capture' and 'playback' required properties: +- stream-name: name of the stream +- formats: list of the supported formats (see below) +- rates: list of the supported rates (see below) +- #channels: minimum and maximum numbers of channels + +Formats: + "s8" + "u8" + "s16_le" + "s16_be" + "u16_le" + "u16_be" + "s24_le" + "s24_be" + "u24_le" + "u24_be" + "s32_le" + "s32_be" + "u32_le" + "u32_be" + "float_le" + "float_be" + "float64_le" + "float64_be" + "iec958_subframe_le" + "iec958_subframe_be" + "mu_law" + "a_law" + "ima_adpcm" + "mpeg" + "gsm" + "special" + "s24_3l" + "s24_3be" + "u24_3le" + "u24_3be" + "s20_3le" + "s20_3be" + "u20_3le" + "u20_3be" + "s18_3le" + "s18_3b" + "u18_3le" + "u18_3be" + "g723_24" + "g723_24_1b" + "g723_40" + "g723_40_1b" + "dsd_u8" + "dsd_u16_le" + +Rates: + 5512 + 8000 + 11025 + 16000 + 22050 + 32000 + 44100 + 48000 + 64000 + 88200 + 96000 + 176400 + 192000 + +Examples: + + spdifo: spdif-transmitter { + compatible = "linux,simple-codec"; + dai-name = "dit-hifi"; + playback { + stream-name = "S/PDIF Playback"; + formats = "s16_le"; + rates = <8000 11025 16000 22050 32000 44100 + 48000 64000 88200 96000>; + #channels = <1 384>; + }; + }; + + hdmio: hdmi-transmitter { + compatible = "linux,simple-codec"; + dai-name = "hdmi-hifi"; + playback { + stream-name = "HDMI Playback"; + formats = "s16_le", "s24_le"; + rates = <32000 44100 48000 88200 96000 + 176400 192000>; + #channels = <2 8>; + }; + }; diff --git a/sound/soc/generic/Kconfig b/sound/soc/generic/Kconfig index 610f612..d9221539 100644 --- a/sound/soc/generic/Kconfig +++ b/sound/soc/generic/Kconfig @@ -2,3 +2,9 @@ config SND_SIMPLE_CARD tristate "ASoC Simple sound card support" help This option enables generic simple sound card support + +config SND_SIMPLE_CODEC + tristate "ASoC Simple sound codec support" + depends on OF + help + This option enables generic simple sound codec support diff --git a/sound/soc/generic/Makefile b/sound/soc/generic/Makefile index 9c3b246..f94620e 100644 --- a/sound/soc/generic/Makefile +++ b/sound/soc/generic/Makefile @@ -1,3 +1,5 @@ snd-soc-simple-card-objs := simple-card.o +snd-soc-simple-codec-objs := simple-codec.o obj-$(CONFIG_SND_SIMPLE_CARD) += snd-soc-simple-card.o +obj-$(CONFIG_SND_SIMPLE_CODEC) += snd-soc-simple-codec.o diff --git a/sound/soc/generic/simple-codec.c b/sound/soc/generic/simple-codec.c new file mode 100644 index 0000000..217cd0e --- /dev/null +++ b/sound/soc/generic/simple-codec.c @@ -0,0 +1,197 @@ +/* + * ALSA SoC simple codec driver + * + * This driver is used by controllers which do not need any codec + * (s/pdif, hdmi..). + * Its parameters are built from a DT description. + * + * Copyright: (C) 2013 Jean-François Moine <moinejf(a)free.fr> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <sound/soc.h> +#include <linux/of.h> + +#define DRV_NAME "simple-codec" + +static struct snd_soc_codec_driver soc_codec_simple_codec; + +static char *format_tb[] = { + "s8", "u8", "s16_le", "s16_be", + "u16_le", "u16_be", "s24_le", "s24_be", + "u24_le", "u24_be", "s32_le", "s32_be", + "u32_le", "u32_be", "float_le", "float_be", + + "float64_le", "float64_be", "iec958_subframe_le", "iec958_subframe_be", + "mu_law", "a_law", "ima_adpcm", "mpeg", + "gsm", "", "", "", + "", "", "", "special", + + "s24_3l", "s24_3be", "u24_3le", "u24_3be", + "s20_3le", "s20_3be", "u20_3le", "u20_3be", + "s18_3le", "s18_3b", "u18_3le", "u18_3be", + "g723_24", "g723_24_1b", "g723_40", "g723_40_1b", + + "dsd_u8", "dsd_u16_le", +}; + +#if SNDRV_PCM_RATE_5512 != 1 << 0 || SNDRV_PCM_RATE_192000 != 1 << 12 +#error "Change this table" +#endif +static u32 rate_tb[] = { 5512, 8000, 11025, 16000, 22050, 32000, 44100, + 48000, 64000, 88200, 96000, 176400, 192000 }; + +static int get_format(u64 *p_formats, char *p) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(format_tb); i++) { + if (strcmp(format_tb[i], p) == 0) { + *p_formats |= 1 << i; + return 0; + } + } + return -EINVAL; +} + +static int get_rate(u32 *p_rates, u32 val) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rate_tb); i++) { + if (rate_tb[i] == val) { + *p_rates |= 1 << i; + return 0; + } + } + return -EINVAL; +} + +static int stream_populate(struct device *dev, + struct snd_soc_pcm_stream *stream, + struct device_node *np) +{ + u64 formats; + u32 rates, val; + char *p; + int i, ret; + + ret = of_property_read_string(np, "stream-name", &stream->stream_name); + if (ret) { + dev_err(dev, "Failed to parse stream-name string\n"); + return ret; + } + + i = 0; + formats = 0; + for (;;) { + ret = of_property_read_string_index(np, "formats", i, + (const char **) &p); + if (ret) + break; + ret = get_format(&formats, p); + if (ret) { + dev_err(dev, "Bad codec format\n"); + return ret; + } + i++; + } + stream->formats = formats; + + i = 0; + rates = 0; + for (;;) { + ret = of_property_read_u32_index(np, "rates", i, &val); + if (ret) + break; + ret = get_rate(&rates, val); + if (ret) { + dev_err(dev, "Bad rate\n"); + return ret; + } + i++; + } + stream->rates = rates; + + ret = of_property_read_u32_index(np, "#channels", 0, &val); + if (ret) { + dev_err(dev, "Bad min channel\n"); + return ret; + } + stream->channels_min = val; + ret = of_property_read_u32_index(np, "#channels", 1, &val); + if (ret) { + dev_err(dev, "Bad max channel\n"); + return ret; + } + stream->channels_max = val; + + return 0; +} + +static int simple_codec_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct device_node *np = dev->of_node; + struct device_node *np2; + struct snd_soc_dai_driver *dai; + int ret; + + dai = devm_kzalloc(dev, sizeof *dai, GFP_KERNEL); + if (!dai) + return -ENOMEM; + + ret = of_property_read_string(np, "dai-name", &dai->name); + if (ret) { + dev_err(dev, "Failed to parse dai-name\n"); + return ret; + } + + np2 = of_find_node_by_name(np, "capture"); + if (np2) { + ret = stream_populate(dev, &dai->capture, np2); + if (ret) + return ret; + } + np2 = of_find_node_by_name(np, "playback"); + if (np2) { + ret = stream_populate(dev, &dai->playback, np2); + if (ret) + return ret; + } + + return snd_soc_register_codec(dev, &soc_codec_simple_codec, + dai, 1); +} + +static int simple_codec_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static const struct of_device_id simple_codec_of_match[] = { + { .compatible = "linux,simple-codec", }, + { } +}; +MODULE_DEVICE_TABLE(of, simple_codec_of_match); + +static struct platform_driver simple_codec_driver = { + .probe = simple_codec_probe, + .remove = simple_codec_remove, + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .of_match_table = simple_codec_of_match, + }, +}; + +module_platform_driver(simple_codec_driver); + +MODULE_AUTHOR("Jean-Francois Moine <moinejf(a)free.fr>"); +MODULE_DESCRIPTION("simple codec driver"); +MODULE_LICENSE("GPL"); -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

5 12

[alsa-devel] [PATCH] ASoC: pcm: Remove extra spaces from dev_ prints
by Jarkko Nikula 30 Sep '13

30 Sep '13

dev_ prints are already prefixed by ": " before format string so there is no need for extra spaces. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/soc-pcm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 330c9a6..d449872 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -721,7 +721,7 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, list_add(&dpcm->list_be, &fe->dpcm[stream].be_clients); list_add(&dpcm->list_fe, &be->dpcm[stream].fe_clients); - dev_dbg(fe->dev, " connected new DPCM %s path %s %s %s\n", + dev_dbg(fe->dev, "connected new DPCM %s path %s %s %s\n", stream ? "capture" : "playback", fe->dai_link->name, stream ? "<-" : "->", be->dai_link->name); @@ -749,7 +749,7 @@ static void dpcm_be_reparent(struct snd_soc_pcm_runtime *fe, if (dpcm->fe == fe) continue; - dev_dbg(fe->dev, " reparent %s path %s %s %s\n", + dev_dbg(fe->dev, "reparent %s path %s %s %s\n", stream ? "capture" : "playback", dpcm->fe->dai_link->name, stream ? "<-" : "->", dpcm->be->dai_link->name); @@ -773,7 +773,7 @@ static void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) if (dpcm->state != SND_SOC_DPCM_LINK_STATE_FREE) continue; - dev_dbg(fe->dev, " freed DSP %s path %s %s %s\n", + dev_dbg(fe->dev, "freed DSP %s path %s %s %s\n", stream ? "capture" : "playback", fe->dai_link->name, stream ? "<-" : "->", dpcm->be->dai_link->name); @@ -2116,7 +2116,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) pcm->private_free = platform->driver->pcm_free; out: - dev_info(rtd->card->dev, " %s <-> %s mapping ok\n", codec_dai->name, + dev_info(rtd->card->dev, "%s <-> %s mapping ok\n", codec_dai->name, cpu_dai->name); return ret; } -- 1.8.4.rc3

2 1

[alsa-devel] [RFC] ALSA: Reduce delay for a blocking pcm_drain.
by Dylan Reid 30 Sep '13

30 Sep '13

This patch addresses two issues related to calling snd_pcm_drain. If no_period_wakeup is set, then snd_pcm_drain would wait MAX_SCHEDULE_TIMEOUT jiffies, without a wakeup pending this will leave the calling task blocked indefinitely. Instead block for double the buffer size played back at a low sample rate (4k chosen arbitrarily). Also if the stream is running with period wakeups but with a long period, the delay could be seconds. If only a small part of the buffer is being used, this is unnecessary. Instead wait for the remaining samples to play out, plus one millisecond. This allows systems that fill the buffer only part way to still do a blocking drain before closing to avoid losing samples. Change-Id: I84ee1a52d9bdf80ea6065fb9533e565d3e6c8021 Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- This can be worked around in user space as well by using a nonblocking drain and sleeping based on how many samples are queued. Would that be the right thing to do or is this (or something similar) worth fixing here. To test this I opened a pcm and set no period wakeups, played a few samples and then called drain, which hung indefinitely. Using a USB headset with a max period size of 128k but only 15ms of the buffer filled, having the drain time shorter without losing samples makes a blocking call feasible. --- sound/core/pcm_native.c | 17 +++++++---------- 1 file changed, 7 insertions(+), 10 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index f4aaf5a..a5f0695 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1481,7 +1481,7 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, } for (;;) { - long tout; + long tout, tout_ms; struct snd_pcm_runtime *to_check; if (signal_pending(current)) { result = -ERESTARTSYS; @@ -1505,16 +1505,13 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, snd_pcm_stream_unlock_irq(substream); up_read(&snd_pcm_link_rwsem); snd_power_unlock(card); - if (runtime->no_period_wakeup) - tout = MAX_SCHEDULE_TIMEOUT; - else { - tout = 10; - if (runtime->rate) { - long t = runtime->period_size * 2 / runtime->rate; - tout = max(t, tout); - } - tout = msecs_to_jiffies(tout * 1000); + tout_ms = runtime->buffer_size * 2 * 1000 / 4000; + if (runtime->rate) { + snd_pcm_sframes_t hw_avail; + hw_avail = snd_pcm_playback_hw_avail(runtime); + tout_ms = 1 + hw_avail * 1000 / runtime->rate; } + tout = msecs_to_jiffies(tout_ms); tout = schedule_timeout_interruptible(tout); snd_power_lock(card); down_read(&snd_pcm_link_rwsem); -- 1.8.1.3.605.g02339dd

3 3

[alsa-devel] [PATCH] Enable internal mic on a Thinkpad machine
by David Henningsson 30 Sep '13

30 Sep '13

From: Kailang <kailang(a)realtek.com> More thorough testing showed that these verbs were necessary to improve quality of the internal mic. Patch originally from Realtek. Cc: stable(a)kernel.org (3.11) BugLink: https://bugs.launchpad.net/bugs/1231931 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bc07d36..c4d54a9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3531,6 +3531,7 @@ enum { ALC269VB_FIXUP_ORDISSIMO_EVE2, ALC283_FIXUP_CHROME_BOOK, ALC282_FIXUP_ASUS_TX300, + ALC283_FIXUP_INT_MIC, }; static const struct hda_fixup alc269_fixups[] = { @@ -3790,6 +3791,16 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc282_fixup_asus_tx300, }, + [ALC283_FIXUP_INT_MIC] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + {0x20, AC_VERB_SET_COEF_INDEX, 0x1a}, + {0x20, AC_VERB_SET_PROC_COEF, 0x0011}, + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -3874,7 +3885,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), - SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x5026, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), -- 1.7.9.5

2 2

Re: [alsa-devel] pcm_rate API question
by Rob Sykes 30 Sep '13

30 Sep '13

> From: Rob Sykes <aquegg(a)yahoo.co.uk> >To: "alsa-devel(a)alsa-project.org" <alsa-devel(a)alsa-project.org> >Sent: Friday, 27 September 2013, 16:28 >Subject: Re: [alsa-devel] pcm_rate API question > >> From: Clemens Ladisch <clemens(a)ladisch.de> >> To: Rob Sykes <aquegg(a)yahoo.co.uk>; "alsa-devel(a)alsa-project.org" <alsa-devel(a)alsa-project.org> >> Cc: >> Sent: Friday, 27 September 2013, 13:38 >> Subject: Re: [alsa-devel] pcm_rate API question >> >> Rob Sykes wrote: >>> void (*convert_s16)(void *obj, int16_t *dst, unsigned int >> dst_frames, >>> const int16_t *src, unsigned int src_frames); >>> >>> is the plugin obliged always to consume src_frames and generate dst_frames >> samples? >> >> Yes. src_frames and dst_frames are the period sizes of the respective >> devices; their values will never change, and they define the actual >> conversion ratio. > >Thanks, Clemens. Typically there is some buffering/delay within a resampler -- how does the pcm_rate API allow for this? E.g. from the libsamplerate FAQ: > >Q6 : I'm use the SRC_SINC_* converters and up-sampling by a ratio of 2. I reset the converter and put in 1000 samples and I expect to get 2000 samples out, but I'm getting less than that. Why? >The short answer is that there is a transport delay inside the converter itself. Long answer follows. By way of example, the first time you call src_process() you might only get 1900 samples out. > However, after that first call all subsequent calls will probably get you about 2000 samples out for every 1000 samples you put in. Also, just noticed that when performing a rate conversion from 44100 -> 48000, the buffer sizes used by alsa are 5512 and 6000, however the actual ratio for the required conversion is 5512.5 to 6000. If, as you say above, the buffer sizes define the conversion ratio, then this seems to imply that alsa is unable to perform accurate rate conversion for one of the most common uses! — am I missing something here? A pointer to the API documentation, or perhaps some relevant threads, would be very useful. Cheers, Rob

2 3

Re: [alsa-devel] [RFC] ALSA: Reduce delay for a blocking pcm_drain.
by Raymond Yau 30 Sep '13

30 Sep '13

> > This can be worked around in user space as well by using a nonblocking drain and > sleeping based on how many samples are queued. Would that be the right thing to > do or is this (or something similar) worth fixing here. > > To test this I opened a pcm and set no period wakeups, played a few samples and > then called drain, which hung indefinitely. Using a USB headset with a > max period size of 128k but only 15ms of the buffer filled, having the > drain time shorter without losing samples makes a blocking call > feasible. > Do you mean that you tested on a patched version of USB audio driver since only three drivers support SNDRV_PCM_INFO_NO_PERIOD_WAKEUP http://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/log/sound?qt=gr…

2 1

Re: [alsa-devel] Bug: Conexant CX20588, Kernel 3.11, Win8 changes GPIO settings
by Takashi Iwai 30 Sep '13

30 Sep '13

At Fri, 27 Sep 2013 22:38:22 +0200, Christopher wrote: > > I attached the alsa-info. This issue seems to be specific to the driver > provided by Acer (with thx, equalizer etc.) > Thanks for your help. Thanks. Could you check whether the patch below works? Takashi --- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: hda - Fix GPIO for Acer Aspire 3830TG Acer Aspire 3830TG seems requiring GPIO bit 0 as the primary mute control. When a machine is booted after Windows 8, the GPIO pin is turned off and it results in the silent output. This patch adds the manual fixup of GPIO bit 0 for this model. Reported-by: Christopher <DIDI2002(a)web.de> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_conexant.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 4edd2d0..ec68eae 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3231,6 +3231,7 @@ enum { CXT_FIXUP_INC_MIC_BOOST, CXT_FIXUP_HEADPHONE_MIC_PIN, CXT_FIXUP_HEADPHONE_MIC, + CXT_FIXUP_GPIO1, }; static void cxt_fixup_stereo_dmic(struct hda_codec *codec, @@ -3375,6 +3376,15 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = cxt_fixup_headphone_mic, }, + [CXT_FIXUP_GPIO1] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x01, AC_VERB_SET_GPIO_MASK, 0x01 }, + { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01 }, + { 0x01, AC_VERB_SET_GPIO_DATA, 0x01 }, + { } + }, + }, }; static const struct snd_pci_quirk cxt5051_fixups[] = { @@ -3384,6 +3394,7 @@ static const struct snd_pci_quirk cxt5051_fixups[] = { static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT_FIXUP_GPIO1), SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T410", CXT_PINCFG_LENOVO_TP410), -- 1.8.4

1 0

[alsa-devel] [PATCH 1/2] [RESEND] snd-ac97-codec: Add ID for TI TLV320AIC27 codec
by Ondrej Zary 30 Sep '13

30 Sep '13

Add 0x54584e03 ID for TI TLV320AIC27 AC'97 codec according to datasheet: http://www.ti.com/lit/ds/slas253a/slas253a.pdf The weird thing is that the chip is physically marked 320AD91. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- sound/pci/ac97/ac97_codec.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 445ca48..bf578ba2 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -175,6 +175,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x54524106, 0xffffffff, "TR28026", NULL, NULL }, { 0x54524108, 0xffffffff, "TR28028", patch_tritech_tr28028, NULL }, // added by xin jin [07/09/99] { 0x54524123, 0xffffffff, "TR28602", NULL, NULL }, // only guess --jk [TR28023 = eMicro EM28023 (new CT1297)] +{ 0x54584e03, 0xffffffff, "TLV320AIC27", NULL, NULL }, { 0x54584e20, 0xffffffff, "TLC320AD9xC", NULL, NULL }, { 0x56494161, 0xffffffff, "VIA1612A", NULL, NULL }, // modified ICE1232 with S/PDIF { 0x56494170, 0xffffffff, "VIA1617A", patch_vt1617a, NULL }, // modified VT1616 with S/PDIF -- Ondrej Zary

2 1

[alsa-devel] [PATCH] caiaq: LED support for Maschine Controller
by Hannes Gräuler 30 Sep '13

30 Sep '13

This patch adds LED support for the Native Instruments Maschine Controller. It adds ALSA controls for dimming the LEDs of all buttons and the backlight of the two displays. Signed-off-by: Hannes Gräuler <hgraeule(a)uos.de> --- sound/usb/caiaq/control.c | 92 +++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/caiaq/device.c | 25 +++++++++++++ sound/usb/caiaq/device.h | 5 +++ 3 files changed, 122 insertions(+) diff --git a/sound/usb/caiaq/control.c b/sound/usb/caiaq/control.c index ae6b50f..f65fc09 100644 --- a/sound/usb/caiaq/control.c +++ b/sound/usb/caiaq/control.c @@ -28,6 +28,7 @@ #include "control.h" #define CNT_INTVAL 0x10000 +#define MASCHINE_BANK_SIZE 32 static int control_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) @@ -105,6 +106,10 @@ static int control_put(struct snd_kcontrol *kcontrol, USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1)) cmd = EP1_CMD_DIMM_LEDS; + if (cdev->chip.usb_id == + USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER)) + cmd = EP1_CMD_DIMM_LEDS; + if (pos & CNT_INTVAL) { int i = pos & ~CNT_INTVAL; @@ -121,6 +126,20 @@ static int control_put(struct snd_kcontrol *kcontrol, usb_sndbulkpipe(cdev->chip.dev, 8), cdev->ep8_out_buf, sizeof(cdev->ep8_out_buf), &actual_len, 200); + } else if (cdev->chip.usb_id == + USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER)) { + + int bank = 0; + int offset = 0; + + if (i >= MASCHINE_BANK_SIZE) { + bank = 0x1e; + offset = MASCHINE_BANK_SIZE; + } + + snd_usb_caiaq_send_command_bank(cdev, cmd, bank, + cdev->control_state + offset, + MASCHINE_BANK_SIZE); } else { snd_usb_caiaq_send_command(cdev, cmd, cdev->control_state, sizeof(cdev->control_state)); @@ -490,6 +509,74 @@ static struct caiaq_controller kontrols4_controller[] = { { "LED: FX2: Mode", 133 | CNT_INTVAL }, }; +static struct caiaq_controller maschine_controller[] = { + { "LED: Pad 1", 3 | CNT_INTVAL }, + { "LED: Pad 2", 2 | CNT_INTVAL }, + { "LED: Pad 3", 1 | CNT_INTVAL }, + { "LED: Pad 4", 0 | CNT_INTVAL }, + { "LED: Pad 5", 7 | CNT_INTVAL }, + { "LED: Pad 6", 6 | CNT_INTVAL }, + { "LED: Pad 7", 5 | CNT_INTVAL }, + { "LED: Pad 8", 4 | CNT_INTVAL }, + { "LED: Pad 9", 11 | CNT_INTVAL }, + { "LED: Pad 10", 10 | CNT_INTVAL }, + { "LED: Pad 11", 9 | CNT_INTVAL }, + { "LED: Pad 12", 8 | CNT_INTVAL }, + { "LED: Pad 13", 15 | CNT_INTVAL }, + { "LED: Pad 14", 14 | CNT_INTVAL }, + { "LED: Pad 15", 13 | CNT_INTVAL }, + { "LED: Pad 16", 12 | CNT_INTVAL }, + + { "LED: Mute", 16 | CNT_INTVAL }, + { "LED: Solo", 17 | CNT_INTVAL }, + { "LED: Select", 18 | CNT_INTVAL }, + { "LED: Duplicate", 19 | CNT_INTVAL }, + { "LED: Navigate", 20 | CNT_INTVAL }, + { "LED: Pad Mode", 21 | CNT_INTVAL }, + { "LED: Pattern", 22 | CNT_INTVAL }, + { "LED: Scene", 23 | CNT_INTVAL }, + + { "LED: Shift", 24 | CNT_INTVAL }, + { "LED: Erase", 25 | CNT_INTVAL }, + { "LED: Grid", 26 | CNT_INTVAL }, + { "LED: Right Bottom", 27 | CNT_INTVAL }, + { "LED: Rec", 28 | CNT_INTVAL }, + { "LED: Play", 29 | CNT_INTVAL }, + { "LED: Left Bottom", 32 | CNT_INTVAL }, + { "LED: Restart", 33 | CNT_INTVAL }, + + { "LED: Group A", 41 | CNT_INTVAL }, + { "LED: Group B", 40 | CNT_INTVAL }, + { "LED: Group C", 37 | CNT_INTVAL }, + { "LED: Group D", 36 | CNT_INTVAL }, + { "LED: Group E", 39 | CNT_INTVAL }, + { "LED: Group F", 38 | CNT_INTVAL }, + { "LED: Group G", 35 | CNT_INTVAL }, + { "LED: Group H", 34 | CNT_INTVAL }, + + { "LED: Auto Write", 42 | CNT_INTVAL }, + { "LED: Snap", 43 | CNT_INTVAL }, + { "LED: Right Top", 44 | CNT_INTVAL }, + { "LED: Left Top", 45 | CNT_INTVAL }, + { "LED: Sampling", 46 | CNT_INTVAL }, + { "LED: Browse", 47 | CNT_INTVAL }, + { "LED: Step", 48 | CNT_INTVAL }, + { "LED: Control", 49 | CNT_INTVAL }, + + { "LED: Top Button 1", 57 | CNT_INTVAL }, + { "LED: Top Button 2", 56 | CNT_INTVAL }, + { "LED: Top Button 3", 55 | CNT_INTVAL }, + { "LED: Top Button 4", 54 | CNT_INTVAL }, + { "LED: Top Button 5", 53 | CNT_INTVAL }, + { "LED: Top Button 6", 52 | CNT_INTVAL }, + { "LED: Top Button 7", 51 | CNT_INTVAL }, + { "LED: Top Button 8", 50 | CNT_INTVAL }, + + { "LED: Note Repeat", 58 | CNT_INTVAL }, + + { "Backlight Display", 59 | CNT_INTVAL } +}; + static int add_controls(struct caiaq_controller *c, int num, struct snd_usb_caiaqdev *cdev) { @@ -553,6 +640,11 @@ int snd_usb_caiaq_control_init(struct snd_usb_caiaqdev *cdev) ret = add_controls(kontrols4_controller, ARRAY_SIZE(kontrols4_controller), cdev); break; + + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER): + ret = add_controls(maschine_controller, + ARRAY_SIZE(maschine_controller), cdev); + break; } return ret; diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 1a61dd1..bc55f70 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -235,6 +235,31 @@ int snd_usb_caiaq_send_command(struct snd_usb_caiaqdev *cdev, cdev->ep1_out_buf, len+1, &actual_len, 200); } +int snd_usb_caiaq_send_command_bank(struct snd_usb_caiaqdev *cdev, + unsigned char command, + unsigned char bank, + const unsigned char *buffer, + int len) +{ + int actual_len; + struct usb_device *usb_dev = cdev->chip.dev; + + if (!usb_dev) + return -EIO; + + if (len > EP1_BUFSIZE - 2) + len = EP1_BUFSIZE - 2; + + if (buffer && len > 0) + memcpy(cdev->ep1_out_buf+2, buffer, len); + + cdev->ep1_out_buf[0] = command; + cdev->ep1_out_buf[1] = bank; + + return usb_bulk_msg(usb_dev, usb_sndbulkpipe(usb_dev, 1), + cdev->ep1_out_buf, len+2, &actual_len, 200); +} + int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *cdev, int rate, int depth, int bpp) { diff --git a/sound/usb/caiaq/device.h b/sound/usb/caiaq/device.h index ad102fa..ab0f752 100644 --- a/sound/usb/caiaq/device.h +++ b/sound/usb/caiaq/device.h @@ -128,5 +128,10 @@ int snd_usb_caiaq_send_command(struct snd_usb_caiaqdev *cdev, unsigned char command, const unsigned char *buffer, int len); +int snd_usb_caiaq_send_command_bank(struct snd_usb_caiaqdev *cdev, + unsigned char command, + unsigned char bank, + const unsigned char *buffer, + int len); #endif /* CAIAQ_DEVICE_H */ -- 1.8.4

3 2

[alsa-devel] ASoC updates for v3.12
by Mark Brown 30 Sep '13

30 Sep '13

The following changes since commit 4a10c2ac2f368583138b774ca41fac4207911983: Linux 3.12-rc2 (2013-09-23 15:41:09 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.12-rc2 for you to fetch changes up to 267666ea3bd6e804c2c8bbafbe628026be963d06: Merge remote-tracking branch 'asoc/fix/max98095' into asoc-linus (2013-09-27 14:56:24 +0100) ---------------------------------------------------------------- ASoC: Fixes for v3.12 Nothing too exciting here, all driver specific except for the fix from Liam for DPCM systems which have both front and back end DAIs which is not yet used by anything in mainline. ---------------------------------------------------------------- Dan Carpenter (3): ASoC: max98095: a couple array underflows ASoC: ab8500-codec: info leak in anc_status_control_put() ASoC: 88pm860x: array overflow in snd_soc_put_volsw_2r_st() Liam Girdwood (1): ASoC: core: Only add platform DAI widgets once. Mark Brown (6): Merge remote-tracking branch 'asoc/fix/88pm860x' into asoc-linus Merge remote-tracking branch 'asoc/fix/ab8500' into asoc-linus Merge remote-tracking branch 'asoc/fix/bfin' into asoc-linus Merge remote-tracking branch 'asoc/fix/core' into asoc-linus Merge remote-tracking branch 'asoc/fix/fsl' into asoc-linus Merge remote-tracking branch 'asoc/fix/max98095' into asoc-linus Philipp Zabel (2): ASoC: imx-sgtl5000: do not use devres on a foreign device ASoC: imx-sgtl5000: Fix uninitialized pointer use in error path Valentin Ilie (1): ASoC: blackfin: Add missing break statement to bf6xx sound/soc/blackfin/bf6xx-i2s.c | 1 + sound/soc/codecs/88pm860x-codec.c | 3 +++ sound/soc/codecs/ab8500-codec.c | 7 ++++++- sound/soc/codecs/max98095.c | 4 ++-- sound/soc/fsl/imx-sgtl5000.c | 7 +++++-- sound/soc/soc-core.c | 1 - 6 files changed, 17 insertions(+), 6 deletions(-)

2 1

Re: [alsa-devel] Realtek ALC286 SONY Vaio Pro 13 Ubuntu 13.10
by David Henningsson 30 Sep '13

30 Sep '13

(Adding alsa-devel to CC) On 09/28/2013 05:53 AM, Yuval Levy wrote: > Sirs: > > I have been pointed to your patch to implement headset jack > functionality for some Dell hw [1] after reporting and investigating > soud support issues [2] with a SONY Vaio Pro 13 (Realtek ALC286). This different headset mode support is likely unrelated to your problem. And they don't cover your codec (ALC286), so you would need another set of verbs too, things that only Realtek knows. > Output to internal loudspeakers and HDMI works as expected, but neither > internal microphone nor the headset work (and they do work in Windows, > so it is not a hardware issue). An important distinction is here: to they work in Windows without specific Realtek drivers installed, or do you require Realtek's Windows drivers for this support? Second, to do some actual triaging. I see you have been running hda-jack-sense-test. Could you run it all three scenarios: 1) nothing connected 2) headphone (without mic) connected 3) headset (heaphone + mic combo) connected ...and submit the result? In ideal circumstances, 0x21 should report "Yes" in scenario 2 and 3, and 0x18 should only report yes in scenario 3. If this is not the result you get, try using hda-jack-retask to turn off the jack detection: in Ubuntu 13.10 just install the alsa-tools-gui package (no need to use my ppa). Click "advanced override", find pins 0x18 or 0x21, change "jack detection" from "yes" to "no" and click apply. Now you should be able to manually choose between internal and external microphone, as well as between headphones as speakers. See if this helps. > As you can see in the bug thread, I have been helped in the diagnosis by > a very friendly expert, but it seems that we have reached the limit and > without support from Realtek or Sony the issue will go unresolved. I'm afraid Sony is not a company that actively supports Linux for the desktop machines, sorry. Canonical, which seems to have the most contracts for preloading Linux these days, has a list of certified hardware here: http://www.ubuntu.com/certification/ > Would you kindly help me find out what need to be done for the Realtek > ALC286 in the SONY Vaio Pro 13 to be supported in Ubuntu Linux? > > Many thanks, > Yuv > > [1] > <https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p…> > [2] <https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/1228689> > -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

1 0

[alsa-devel] [PATCH] ASoC: si476x: Remove custom register I/O implementation
by Mark Brown 28 Sep '13

28 Sep '13

From: Mark Brown <broonie(a)linaro.org> The current si476x I/O implementation wraps the regmap for the core with functions that make the register map cache only when the device is powered down. This implementation appears to be incomplete since there is no code to synchronise the cache so writes done while the core is powered down will be ignored, the device will only be configured if it is powered. A better and more idiomatic approach would be to have the MFD manage the cache, making the device cache only when it powers things down. This also allows ASoC to use the standard regmap helpers for the device which helps remove the ASoC custom ones so do convert to do that. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/si476x.c | 46 +--------------------------------------------- 1 file changed, 1 insertion(+), 45 deletions(-) diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index 38f3b10..ec29d26 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -60,48 +60,6 @@ enum si476x_pcm_format { SI476X_PCM_FORMAT_S24_LE = 6, }; -static unsigned int si476x_codec_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - int err; - unsigned int val; - struct si476x_core *core = codec->control_data; - - si476x_core_lock(core); - if (!si476x_core_is_powered_up(core)) - regcache_cache_only(core->regmap, true); - - err = regmap_read(core->regmap, reg, &val); - - if (!si476x_core_is_powered_up(core)) - regcache_cache_only(core->regmap, false); - si476x_core_unlock(core); - - if (err < 0) - return err; - - return val; -} - -static int si476x_codec_write(struct snd_soc_codec *codec, - unsigned int reg, unsigned int val) -{ - int err; - struct si476x_core *core = codec->control_data; - - si476x_core_lock(core); - if (!si476x_core_is_powered_up(core)) - regcache_cache_only(core->regmap, true); - - err = regmap_write(core->regmap, reg, val); - - if (!si476x_core_is_powered_up(core)) - regcache_cache_only(core->regmap, false); - si476x_core_unlock(core); - - return err; -} - static const struct snd_soc_dapm_widget si476x_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("LOUT"), SND_SOC_DAPM_OUTPUT("ROUT"), @@ -239,7 +197,7 @@ static int si476x_codec_hw_params(struct snd_pcm_substream *substream, static int si476x_codec_probe(struct snd_soc_codec *codec) { - codec->control_data = i2c_mfd_cell_to_core(codec->dev); + codec->control_data = dev_get_regmap(codec->dev.parent); return 0; } @@ -268,8 +226,6 @@ static struct snd_soc_dai_driver si476x_dai = { static struct snd_soc_codec_driver soc_codec_dev_si476x = { .probe = si476x_codec_probe, - .read = si476x_codec_read, - .write = si476x_codec_write, .dapm_widgets = si476x_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(si476x_dapm_widgets), .dapm_routes = si476x_dapm_routes, -- 1.8.4.rc3

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[alsa-devel] abdul, please add me to your LinkedIn network
by Maen Suleiman via LinkedIn 27 Sep '13

27 Sep '13

LinkedIn ------------ Maen Suleiman has indicated you are a Friend ------------------------------------------ Since you are a person I trust, I wanted to invite you to join my network on LinkedIn. Accept invitation from Maen Suleiman http://www.linkedin.com/e/jpxkk3-hm3ttpo1-7/uii-rtptP1H4ViCg9YiBUOvto1r4_5G… View profile of Maen Suleiman http://www.linkedin.com/e/jpxkk3-hm3ttpo1-7/rso/2181468/NG8u/name/138036396… ------------------------------------------ You are receiving Invitation emails. This email was intended for abdul moiz. Learn why this is included: http://www.linkedin.com/e/jpxkk3-hm3ttpo1-7/plh/http%3A%2F%2Fhelp%2Elinkedi… (c) 2012, LinkedIn Corporation. 2029 Stierlin Ct, Mountain View, CA 94043, USA.

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[alsa-devel] [PATCH] ASoc: kirkwood: Extend the min and max number of bytes per period
by Jean-Francois Moine 27 Sep '13

27 Sep '13

This patch extends the min and max number of bytes per period. It mainly permits to reduce the sound delay in MIDI real-time playing. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/kirkwood/kirkwood.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood.h b/sound/soc/kirkwood/kirkwood.h index f8e1ccc..bf23afb 100644 --- a/sound/soc/kirkwood/kirkwood.h +++ b/sound/soc/kirkwood/kirkwood.h @@ -123,8 +123,8 @@ /* need to find where they come from */ #define KIRKWOOD_SND_MIN_PERIODS 8 #define KIRKWOOD_SND_MAX_PERIODS 16 -#define KIRKWOOD_SND_MIN_PERIOD_BYTES 0x4000 -#define KIRKWOOD_SND_MAX_PERIOD_BYTES 0x4000 +#define KIRKWOOD_SND_MIN_PERIOD_BYTES 0x800 +#define KIRKWOOD_SND_MAX_PERIOD_BYTES 0x8000 #define KIRKWOOD_SND_MAX_BUFFER_BYTES (KIRKWOOD_SND_MAX_PERIOD_BYTES \ * KIRKWOOD_SND_MAX_PERIODS) -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

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[alsa-devel] usb-audio mic problem with Polycom C100S
by Meelis Roos 27 Sep '13

27 Sep '13

I tried to use Skype (4.1) with Polycom Communicator C100S. Audio output works fine but mic seems to be dead, although it is set to capture and volume is even max. Dmesg tells the following two kinds of "cannot get freq at ep ..." messages while trying to use it. Linux is 3.12-rc2 but I saw the same behaviour a couple releases ago when I first tried it. I am not subscribed to alsa-devel, please CC in replies. dmesg [117273.437015] usb 1-1.2.3: new full-speed USB device number 9 using ehci-pci [117273.901033] usb 1-1.2.3: New USB device found, idVendor=095d, idProduct=0005 [117273.901036] usb 1-1.2.3: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [117273.901037] usb 1-1.2.3: Product: Polycom Communicator [117273.901038] usb 1-1.2.3: Manufacturer: Polycom,Inc. [117273.901039] usb 1-1.2.3: SerialNumber: L80646021E21 [117273.962134] hid-generic 0003:095D:0005.0005: hiddev0,hidraw3: USB HID v1.00 Device [Polycom,Inc. Polycom Communicator] on usb-0000:00:1a.0-1.2.3/input3 [117273.974525] ALSA sound/usb/clock.c:309 9:2:1: cannot get freq at ep 0x1 [117273.978540] ALSA sound/usb/clock.c:309 9:1:1: cannot get freq at ep 0x82 [117274.019749] usbcore: registered new interface driver snd-usb-audio [117380.633919] ALSA sound/usb/clock.c:309 9:2:1: cannot get freq at ep 0x1 [117380.636540] ALSA sound/usb/clock.c:309 9:2:1: cannot get freq at ep 0x1 [117383.587846] ALSA sound/usb/clock.c:309 9:2:1: cannot get freq at ep 0x1 [117383.619463] ALSA sound/usb/clock.c:309 9:1:1: cannot get freq at ep 0x82 [117383.639484] ALSA sound/usb/clock.c:309 9:2:1: cannot get freq at ep 0x1 [117383.671451] ALSA sound/usb/clock.c:309 9:1:1: cannot get freq at ep 0x82 [117383.697219] ALSA sound/usb/clock.c:309 9:2:1: cannot get freq at ep 0x1 [117383.699840] ALSA sound/usb/clock.c:309 9:2:1: cannot get freq at ep 0x1 [117385.663752] ALSA sound/usb/clock.c:309 9:1:1: cannot get freq at ep 0x82 [117385.666495] ALSA sound/usb/clock.c:309 9:1:1: cannot get freq at ep 0x82 [117386.090400] ALSA sound/usb/clock.c:309 9:2:1: cannot get freq at ep 0x1 [117386.122894] ALSA sound/usb/clock.c:309 9:1:1: cannot get freq at ep 0x82 [117386.143015] ALSA sound/usb/clock.c:309 9:2:1: cannot get freq at ep 0x1 [117386.174880] ALSA sound/usb/clock.c:309 9:1:1: cannot get freq at ep 0x82 [117386.200626] ALSA sound/usb/clock.c:309 9:2:1: cannot get freq at ep 0x1 [117386.203252] ALSA sound/usb/clock.c:309 9:2:1: cannot get freq at ep 0x1 [117392.696024] ALSA sound/usb/clock.c:309 9:2:1: cannot get freq at ep 0x1 [117392.698650] ALSA sound/usb/clock.c:309 9:2:1: cannot get freq at ep 0x1 [117395.407411] ALSA sound/usb/clock.c:309 9:1:1: cannot get freq at ep 0x82 [117395.410160] ALSA sound/usb/clock.c:309 9:1:1: cannot get freq at ep 0x82 [117449.635703] ALSA sound/usb/clock.c:309 9:2:1: cannot get freq at ep 0x1 [117449.638451] ALSA sound/usb/clock.c:309 9:2:1: cannot get freq at ep 0x1 lsusb -v Bus 001 Device 008: ID 095d:0005 Polycom, Inc. Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.10 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 8 idVendor 0x095d Polycom, Inc. idProduct 0x0005 bcdDevice 1.00 iManufacturer 1 iProduct 2 iSerial 3 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 231 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 500mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 83 bInCollection 2 baInterfaceNr( 0) 2 baInterfaceNr( 1) 1 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0201 Microphone bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 2 bSourceID 1 bControlSize 1 bmaControls( 0) 0x03 Mute Control Volume Control bmaControls( 1) 0x00 bmaControls( 2) 0x00 iFeature 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 4 (MIXER_UNIT) bUnitID 3 bNrInPins 1 baSourceID( 0) 2 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 bmControls 0x00 iMixer 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 4 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 3 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 5 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 6 bSourceID 5 bControlSize 1 bmaControls( 0) 0x03 Mute Control Volume Control bmaControls( 1) 0x00 bmaControls( 2) 0x00 iFeature 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 7 wTerminalType 0x0301 Speaker bAssocTerminal 0 bSourceID 6 iTerminal 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 4 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 1 Discrete tSamFreq[ 0] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 13 Transfer Type Isochronous Synch Type Synchronous Usage Type Data wMaxPacketSize 0x00c0 1x 192 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 5 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 1 Discrete tSamFreq[ 0] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 13 Transfer Type Isochronous Synch Type Synchronous Usage Type Data wMaxPacketSize 0x00c0 1x 192 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 3 Human Interface Device bInterfaceSubClass 0 No Subclass bInterfaceProtocol 0 None iInterface 0 HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 1.00 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 52 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0008 1x 8 bytes bInterval 10 -- Meelis Roos (mroos(a)linux.ee)

2 4

[alsa-devel] [PATCH] ASoC: adav80x: Convert to direct regmap usage
by Lars-Peter Clausen 27 Sep '13

27 Sep '13

Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/adav80x.c | 147 +++++++++++++++++++++++++++++++-------------- 1 file changed, 102 insertions(+), 45 deletions(-) diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index 15b012d0..14a7c16 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -115,22 +115,34 @@ #define ADAV80X_PLL_OUTE_SYSCLKPD(x) BIT(2 - (x)) -static u8 adav80x_default_regs[] = { - 0x00, 0x00, 0x00, 0x00, 0x01, 0x01, 0x02, 0x01, 0x80, 0x26, 0x00, 0x00, - 0x02, 0x40, 0x20, 0x00, 0x09, 0x08, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x04, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0xd1, 0x92, 0xb1, 0x37, - 0x48, 0xd2, 0xfb, 0xca, 0xd2, 0x15, 0xe8, 0x29, 0xb9, 0x6a, 0xda, 0x2b, - 0xb7, 0xc0, 0x11, 0x65, 0x5c, 0xf6, 0xff, 0x8d, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0xa5, 0x00, 0x00, - 0x00, 0xe8, 0x46, 0xe1, 0x5b, 0xd3, 0x43, 0x77, 0x93, 0xa7, 0x44, 0xee, - 0x32, 0x12, 0xc0, 0x11, 0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0x3f, 0x3f, - 0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0x00, 0x1d, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x52, 0x00, +static struct reg_default adav80x_reg_defaults[] = { + { ADAV80X_PLAYBACK_CTRL, 0x01 }, + { ADAV80X_AUX_IN_CTRL, 0x01 }, + { ADAV80X_REC_CTRL, 0x02 }, + { ADAV80X_AUX_OUT_CTRL, 0x01 }, + { ADAV80X_DPATH_CTRL1, 0xc0 }, + { ADAV80X_DPATH_CTRL2, 0x11 }, + { ADAV80X_DAC_CTRL1, 0x00 }, + { ADAV80X_DAC_CTRL2, 0x00 }, + { ADAV80X_DAC_CTRL3, 0x00 }, + { ADAV80X_DAC_L_VOL, 0xff }, + { ADAV80X_DAC_R_VOL, 0xff }, + { ADAV80X_PGA_L_VOL, 0x00 }, + { ADAV80X_PGA_R_VOL, 0x00 }, + { ADAV80X_ADC_CTRL1, 0x00 }, + { ADAV80X_ADC_CTRL2, 0x00 }, + { ADAV80X_ADC_L_VOL, 0xff }, + { ADAV80X_ADC_R_VOL, 0xff }, + { ADAV80X_PLL_CTRL1, 0x00 }, + { ADAV80X_PLL_CTRL2, 0x00 }, + { ADAV80X_ICLK_CTRL1, 0x00 }, + { ADAV80X_ICLK_CTRL2, 0x00 }, + { ADAV80X_PLL_CLK_SRC, 0x00 }, + { ADAV80X_PLL_OUTE, 0x00 }, }; struct adav80x { - enum snd_soc_control_type control_type; + struct regmap *regmap; enum adav80x_clk_src clk_src; unsigned int sysclk; @@ -298,7 +310,7 @@ static int adav80x_set_deemph(struct snd_soc_codec *codec) val = ADAV80X_DAC_CTRL2_DEEMPH_NONE; } - return snd_soc_update_bits(codec, ADAV80X_DAC_CTRL2, + return regmap_update_bits(adav80x->regmap, ADAV80X_DAC_CTRL2, ADAV80X_DAC_CTRL2_DEEMPH_MASK, val); } @@ -394,10 +406,11 @@ static int adav80x_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][0], + regmap_update_bits(adav80x->regmap, adav80x_port_ctrl_regs[dai->id][0], ADAV80X_CAPTURE_MODE_MASK | ADAV80X_CAPTURE_MODE_MASTER, capture); - snd_soc_write(codec, adav80x_port_ctrl_regs[dai->id][1], playback); + regmap_write(adav80x->regmap, adav80x_port_ctrl_regs[dai->id][1], + playback); adav80x->dai_fmt[dai->id] = fmt & SND_SOC_DAIFMT_FORMAT_MASK; @@ -407,6 +420,7 @@ static int adav80x_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) static int adav80x_set_adc_clock(struct snd_soc_codec *codec, unsigned int sample_rate) { + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); unsigned int val; if (sample_rate <= 48000) @@ -414,7 +428,7 @@ static int adav80x_set_adc_clock(struct snd_soc_codec *codec, else val = ADAV80X_ADC_CTRL1_MODULATOR_64FS; - snd_soc_update_bits(codec, ADAV80X_ADC_CTRL1, + regmap_update_bits(adav80x->regmap, ADAV80X_ADC_CTRL1, ADAV80X_ADC_CTRL1_MODULATOR_MASK, val); return 0; @@ -423,6 +437,7 @@ static int adav80x_set_adc_clock(struct snd_soc_codec *codec, static int adav80x_set_dac_clock(struct snd_soc_codec *codec, unsigned int sample_rate) { + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); unsigned int val; if (sample_rate <= 48000) @@ -430,7 +445,7 @@ static int adav80x_set_dac_clock(struct snd_soc_codec *codec, else val = ADAV80X_DAC_CTRL2_DIV2 | ADAV80X_DAC_CTRL2_INTERPOL_128FS; - snd_soc_update_bits(codec, ADAV80X_DAC_CTRL2, + regmap_update_bits(adav80x->regmap, ADAV80X_DAC_CTRL2, ADAV80X_DAC_CTRL2_DIV_MASK | ADAV80X_DAC_CTRL2_INTERPOL_MASK, val); @@ -440,6 +455,7 @@ static int adav80x_set_dac_clock(struct snd_soc_codec *codec, static int adav80x_set_capture_pcm_format(struct snd_soc_codec *codec, struct snd_soc_dai *dai, snd_pcm_format_t format) { + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); unsigned int val; switch (format) { @@ -459,7 +475,7 @@ static int adav80x_set_capture_pcm_format(struct snd_soc_codec *codec, return -EINVAL; } - snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][0], + regmap_update_bits(adav80x->regmap, adav80x_port_ctrl_regs[dai->id][0], ADAV80X_CAPTURE_WORD_LEN_MASK, val); return 0; @@ -491,7 +507,7 @@ static int adav80x_set_playback_pcm_format(struct snd_soc_codec *codec, return -EINVAL; } - snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][1], + regmap_update_bits(adav80x->regmap, adav80x_port_ctrl_regs[dai->id][1], ADAV80X_PLAYBACK_MODE_MASK, val); return 0; @@ -554,8 +570,10 @@ static int adav80x_set_sysclk(struct snd_soc_codec *codec, ADAV80X_ICLK_CTRL1_ICLK2_SRC(clk_id); iclk_ctrl2 = ADAV80X_ICLK_CTRL2_ICLK1_SRC(clk_id); - snd_soc_write(codec, ADAV80X_ICLK_CTRL1, iclk_ctrl1); - snd_soc_write(codec, ADAV80X_ICLK_CTRL2, iclk_ctrl2); + regmap_write(adav80x->regmap, ADAV80X_ICLK_CTRL1, + iclk_ctrl1); + regmap_write(adav80x->regmap, ADAV80X_ICLK_CTRL2, + iclk_ctrl2); snd_soc_dapm_sync(&codec->dapm); } @@ -575,10 +593,12 @@ static int adav80x_set_sysclk(struct snd_soc_codec *codec, mask = ADAV80X_PLL_OUTE_SYSCLKPD(clk_id); if (freq == 0) { - snd_soc_update_bits(codec, ADAV80X_PLL_OUTE, mask, mask); + regmap_update_bits(adav80x->regmap, ADAV80X_PLL_OUTE, + mask, mask); adav80x->sysclk_pd[clk_id] = true; } else { - snd_soc_update_bits(codec, ADAV80X_PLL_OUTE, mask, 0); + regmap_update_bits(adav80x->regmap, ADAV80X_PLL_OUTE, + mask, 0); adav80x->sysclk_pd[clk_id] = false; } @@ -650,9 +670,9 @@ static int adav80x_set_pll(struct snd_soc_codec *codec, int pll_id, return -EINVAL; } - snd_soc_update_bits(codec, ADAV80X_PLL_CTRL1, ADAV80X_PLL_CTRL1_PLLDIV, - pll_ctrl1); - snd_soc_update_bits(codec, ADAV80X_PLL_CTRL2, + regmap_update_bits(adav80x->regmap, ADAV80X_PLL_CTRL1, + ADAV80X_PLL_CTRL1_PLLDIV, pll_ctrl1); + regmap_update_bits(adav80x->regmap, ADAV80X_PLL_CTRL2, ADAV80X_PLL_CTRL2_PLL_MASK(pll_id), pll_ctrl2); if (source != adav80x->pll_src) { @@ -661,7 +681,7 @@ static int adav80x_set_pll(struct snd_soc_codec *codec, int pll_id, else pll_src = ADAV80X_PLL_CLK_SRC_PLL_XIN(pll_id); - snd_soc_update_bits(codec, ADAV80X_PLL_CLK_SRC, + regmap_update_bits(adav80x->regmap, ADAV80X_PLL_CLK_SRC, ADAV80X_PLL_CLK_SRC_PLL_MASK(pll_id), pll_src); adav80x->pll_src = source; @@ -675,6 +695,7 @@ static int adav80x_set_pll(struct snd_soc_codec *codec, int pll_id, static int adav80x_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); unsigned int mask = ADAV80X_DAC_CTRL1_PD; switch (level) { @@ -683,10 +704,12 @@ static int adav80x_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - snd_soc_update_bits(codec, ADAV80X_DAC_CTRL1, mask, 0x00); + regmap_update_bits(adav80x->regmap, ADAV80X_DAC_CTRL1, mask, + 0x00); break; case SND_SOC_BIAS_OFF: - snd_soc_update_bits(codec, ADAV80X_DAC_CTRL1, mask, mask); + regmap_update_bits(adav80x->regmap, ADAV80X_DAC_CTRL1, mask, + mask); break; } @@ -780,7 +803,7 @@ static int adav80x_probe(struct snd_soc_codec *codec) int ret; struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); - ret = snd_soc_codec_set_cache_io(codec, 7, 9, adav80x->control_type); + ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); if (ret) { dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); return ret; @@ -791,23 +814,31 @@ static int adav80x_probe(struct snd_soc_codec *codec) snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL2"); /* Power down S/PDIF receiver, since it is currently not supported */ - snd_soc_write(codec, ADAV80X_PLL_OUTE, 0x20); + regmap_write(adav80x->regmap, ADAV80X_PLL_OUTE, 0x20); /* Disable DAC zero flag */ - snd_soc_write(codec, ADAV80X_DAC_CTRL3, 0x6); + regmap_write(adav80x->regmap, ADAV80X_DAC_CTRL3, 0x6); return adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); } static int adav80x_suspend(struct snd_soc_codec *codec) { - return adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF); + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF); + regcache_cache_only(adav80x->regmap, true); + + return ret; } static int adav80x_resume(struct snd_soc_codec *codec) { + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(adav80x->regmap, false); adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - codec->cache_sync = 1; - snd_soc_cache_sync(codec); + regcache_sync(adav80x->regmap); return 0; } @@ -827,10 +858,6 @@ static struct snd_soc_codec_driver adav80x_codec_driver = { .set_pll = adav80x_set_pll, .set_sysclk = adav80x_set_sysclk, - .reg_word_size = sizeof(u8), - .reg_cache_size = ARRAY_SIZE(adav80x_default_regs), - .reg_cache_default = adav80x_default_regs, - .controls = adav80x_controls, .num_controls = ARRAY_SIZE(adav80x_controls), .dapm_widgets = adav80x_dapm_widgets, @@ -839,18 +866,21 @@ static struct snd_soc_codec_driver adav80x_codec_driver = { .num_dapm_routes = ARRAY_SIZE(adav80x_dapm_routes), }; -static int adav80x_bus_probe(struct device *dev, - enum snd_soc_control_type control_type) +static int adav80x_bus_probe(struct device *dev, struct regmap *regmap) { struct adav80x *adav80x; int ret; + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + adav80x = kzalloc(sizeof(*adav80x), GFP_KERNEL); if (!adav80x) return -ENOMEM; + dev_set_drvdata(dev, adav80x); - adav80x->control_type = control_type; + adav80x->regmap = regmap; ret = snd_soc_register_codec(dev, &adav80x_codec_driver, adav80x_dais, ARRAY_SIZE(adav80x_dais)); @@ -868,6 +898,19 @@ static int adav80x_bus_remove(struct device *dev) } #if defined(CONFIG_SPI_MASTER) +static const struct regmap_config adav80x_spi_regmap_config = { + .val_bits = 8, + .pad_bits = 1, + .reg_bits = 7, + .read_flag_mask = 0x01, + + .max_register = ADAV80X_PLL_OUTE, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = adav80x_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(adav80x_reg_defaults), +}; + static const struct spi_device_id adav80x_spi_id[] = { { "adav801", 0 }, { } @@ -876,7 +919,8 @@ MODULE_DEVICE_TABLE(spi, adav80x_spi_id); static int adav80x_spi_probe(struct spi_device *spi) { - return adav80x_bus_probe(&spi->dev, SND_SOC_SPI); + return adav80x_bus_probe(&spi->dev, + devm_regmap_init_spi(spi, &adav80x_spi_regmap_config)); } static int adav80x_spi_remove(struct spi_device *spi) @@ -896,6 +940,18 @@ static struct spi_driver adav80x_spi_driver = { #endif #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static const struct regmap_config adav80x_i2c_regmap_config = { + .val_bits = 8, + .pad_bits = 1, + .reg_bits = 7, + + .max_register = ADAV80X_PLL_OUTE, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = adav80x_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(adav80x_reg_defaults), +}; + static const struct i2c_device_id adav80x_i2c_id[] = { { "adav803", 0 }, { } @@ -905,7 +961,8 @@ MODULE_DEVICE_TABLE(i2c, adav80x_i2c_id); static int adav80x_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { - return adav80x_bus_probe(&client->dev, SND_SOC_I2C); + return adav80x_bus_probe(&client->dev, + devm_regmap_init_i2c(client, &adav80x_i2c_regmap_config)); } static int adav80x_i2c_remove(struct i2c_client *client) -- 1.8.0

2 1

[alsa-devel] [PATCH 1/3] ASoC: adau1373: Convert to direct regmap usage
by Lars-Peter Clausen 27 Sep '13

27 Sep '13

Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/adau1373.c | 295 +++++++++++++++++++++++++++++++++++--------- 1 file changed, 235 insertions(+), 60 deletions(-) diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 1aa10dd..c57c1f8 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -32,6 +32,7 @@ struct adau1373_dai { }; struct adau1373 { + struct regmap *regmap; struct adau1373_dai dais[3]; }; @@ -152,37 +153,172 @@ struct adau1373 { #define ADAU1373_EP_CTRL_MICBIAS1_OFFSET 4 #define ADAU1373_EP_CTRL_MICBIAS2_OFFSET 2 -static const uint8_t adau1373_default_regs[] = { - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x00 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x10 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x20 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00, 0x00, /* 0x30 */ - 0x00, 0x00, 0x00, 0x80, 0x00, 0x01, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x0a, 0x0a, 0x0a, 0x00, /* 0x40 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x08, 0x08, 0x08, 0x00, 0x00, 0x00, 0x00, /* 0x50 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x60 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x70 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x78, 0x18, 0x00, 0x00, 0x00, 0xc0, 0x00, 0x00, /* 0x80 */ - 0x00, 0xc0, 0x88, 0x7a, 0xdf, 0x20, 0x00, 0x00, - 0x78, 0x18, 0x00, 0x00, 0x00, 0xc0, 0x00, 0x00, /* 0x90 */ - 0x00, 0xc0, 0x88, 0x7a, 0xdf, 0x20, 0x00, 0x00, - 0x78, 0x18, 0x00, 0x00, 0x00, 0xc0, 0x00, 0x00, /* 0xa0 */ - 0x00, 0xc0, 0x88, 0x7a, 0xdf, 0x20, 0x00, 0x00, - 0x00, 0x00, 0x00, 0xff, 0xff, 0xff, 0xff, 0xff, /* 0xb0 */ - 0xff, 0xff, 0xff, 0xff, 0xff, 0x1f, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0xc0 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0xd0 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00, /* 0xe0 */ - 0x00, 0x1f, 0x0f, 0x00, 0x00, +static const struct reg_default adau1373_reg_defaults[] = { + { ADAU1373_INPUT_MODE, 0x00 }, + { ADAU1373_AINL_CTRL(0), 0x00 }, + { ADAU1373_AINR_CTRL(0), 0x00 }, + { ADAU1373_AINL_CTRL(1), 0x00 }, + { ADAU1373_AINR_CTRL(1), 0x00 }, + { ADAU1373_AINL_CTRL(2), 0x00 }, + { ADAU1373_AINR_CTRL(2), 0x00 }, + { ADAU1373_AINL_CTRL(3), 0x00 }, + { ADAU1373_AINR_CTRL(3), 0x00 }, + { ADAU1373_LLINE_OUT(0), 0x00 }, + { ADAU1373_RLINE_OUT(0), 0x00 }, + { ADAU1373_LLINE_OUT(1), 0x00 }, + { ADAU1373_RLINE_OUT(1), 0x00 }, + { ADAU1373_LSPK_OUT, 0x00 }, + { ADAU1373_RSPK_OUT, 0x00 }, + { ADAU1373_LHP_OUT, 0x00 }, + { ADAU1373_RHP_OUT, 0x00 }, + { ADAU1373_ADC_GAIN, 0x00 }, + { ADAU1373_LADC_MIXER, 0x00 }, + { ADAU1373_RADC_MIXER, 0x00 }, + { ADAU1373_LLINE1_MIX, 0x00 }, + { ADAU1373_RLINE1_MIX, 0x00 }, + { ADAU1373_LLINE2_MIX, 0x00 }, + { ADAU1373_RLINE2_MIX, 0x00 }, + { ADAU1373_LSPK_MIX, 0x00 }, + { ADAU1373_RSPK_MIX, 0x00 }, + { ADAU1373_LHP_MIX, 0x00 }, + { ADAU1373_RHP_MIX, 0x00 }, + { ADAU1373_EP_MIX, 0x00 }, + { ADAU1373_HP_CTRL, 0x00 }, + { ADAU1373_HP_CTRL2, 0x00 }, + { ADAU1373_LS_CTRL, 0x00 }, + { ADAU1373_EP_CTRL, 0x00 }, + { ADAU1373_MICBIAS_CTRL1, 0x00 }, + { ADAU1373_MICBIAS_CTRL2, 0x00 }, + { ADAU1373_OUTPUT_CTRL, 0x00 }, + { ADAU1373_PWDN_CTRL1, 0x00 }, + { ADAU1373_PWDN_CTRL2, 0x00 }, + { ADAU1373_PWDN_CTRL3, 0x00 }, + { ADAU1373_DPLL_CTRL(0), 0x00 }, + { ADAU1373_PLL_CTRL1(0), 0x00 }, + { ADAU1373_PLL_CTRL2(0), 0x00 }, + { ADAU1373_PLL_CTRL3(0), 0x00 }, + { ADAU1373_PLL_CTRL4(0), 0x00 }, + { ADAU1373_PLL_CTRL5(0), 0x00 }, + { ADAU1373_PLL_CTRL6(0), 0x02 }, + { ADAU1373_DPLL_CTRL(1), 0x00 }, + { ADAU1373_PLL_CTRL1(1), 0x00 }, + { ADAU1373_PLL_CTRL2(1), 0x00 }, + { ADAU1373_PLL_CTRL3(1), 0x00 }, + { ADAU1373_PLL_CTRL4(1), 0x00 }, + { ADAU1373_PLL_CTRL5(1), 0x00 }, + { ADAU1373_PLL_CTRL6(1), 0x02 }, + { ADAU1373_HEADDECT, 0x00 }, + { ADAU1373_ADC_CTRL, 0x00 }, + { ADAU1373_CLK_SRC_DIV(0), 0x00 }, + { ADAU1373_CLK_SRC_DIV(1), 0x00 }, + { ADAU1373_DAI(0), 0x0a }, + { ADAU1373_DAI(1), 0x0a }, + { ADAU1373_DAI(2), 0x0a }, + { ADAU1373_BCLKDIV(0), 0x00 }, + { ADAU1373_BCLKDIV(1), 0x00 }, + { ADAU1373_BCLKDIV(2), 0x00 }, + { ADAU1373_SRC_RATIOA(0), 0x00 }, + { ADAU1373_SRC_RATIOB(0), 0x00 }, + { ADAU1373_SRC_RATIOA(1), 0x00 }, + { ADAU1373_SRC_RATIOB(1), 0x00 }, + { ADAU1373_SRC_RATIOA(2), 0x00 }, + { ADAU1373_SRC_RATIOB(2), 0x00 }, + { ADAU1373_DEEMP_CTRL, 0x00 }, + { ADAU1373_SRC_DAI_CTRL(0), 0x08 }, + { ADAU1373_SRC_DAI_CTRL(1), 0x08 }, + { ADAU1373_SRC_DAI_CTRL(2), 0x08 }, + { ADAU1373_DIN_MIX_CTRL(0), 0x00 }, + { ADAU1373_DIN_MIX_CTRL(1), 0x00 }, + { ADAU1373_DIN_MIX_CTRL(2), 0x00 }, + { ADAU1373_DIN_MIX_CTRL(3), 0x00 }, + { ADAU1373_DIN_MIX_CTRL(4), 0x00 }, + { ADAU1373_DOUT_MIX_CTRL(0), 0x00 }, + { ADAU1373_DOUT_MIX_CTRL(1), 0x00 }, + { ADAU1373_DOUT_MIX_CTRL(2), 0x00 }, + { ADAU1373_DOUT_MIX_CTRL(3), 0x00 }, + { ADAU1373_DOUT_MIX_CTRL(4), 0x00 }, + { ADAU1373_DAI_PBL_VOL(0), 0x00 }, + { ADAU1373_DAI_PBR_VOL(0), 0x00 }, + { ADAU1373_DAI_PBL_VOL(1), 0x00 }, + { ADAU1373_DAI_PBR_VOL(1), 0x00 }, + { ADAU1373_DAI_PBL_VOL(2), 0x00 }, + { ADAU1373_DAI_PBR_VOL(2), 0x00 }, + { ADAU1373_DAI_RECL_VOL(0), 0x00 }, + { ADAU1373_DAI_RECR_VOL(0), 0x00 }, + { ADAU1373_DAI_RECL_VOL(1), 0x00 }, + { ADAU1373_DAI_RECR_VOL(1), 0x00 }, + { ADAU1373_DAI_RECL_VOL(2), 0x00 }, + { ADAU1373_DAI_RECR_VOL(2), 0x00 }, + { ADAU1373_DAC1_PBL_VOL, 0x00 }, + { ADAU1373_DAC1_PBR_VOL, 0x00 }, + { ADAU1373_DAC2_PBL_VOL, 0x00 }, + { ADAU1373_DAC2_PBR_VOL, 0x00 }, + { ADAU1373_ADC_RECL_VOL, 0x00 }, + { ADAU1373_ADC_RECR_VOL, 0x00 }, + { ADAU1373_DMIC_RECL_VOL, 0x00 }, + { ADAU1373_DMIC_RECR_VOL, 0x00 }, + { ADAU1373_VOL_GAIN1, 0x00 }, + { ADAU1373_VOL_GAIN2, 0x00 }, + { ADAU1373_VOL_GAIN3, 0x00 }, + { ADAU1373_HPF_CTRL, 0x00 }, + { ADAU1373_BASS1, 0x00 }, + { ADAU1373_BASS2, 0x00 }, + { ADAU1373_DRC(0) + 0x0, 0x78 }, + { ADAU1373_DRC(0) + 0x1, 0x18 }, + { ADAU1373_DRC(0) + 0x2, 0x00 }, + { ADAU1373_DRC(0) + 0x3, 0x00 }, + { ADAU1373_DRC(0) + 0x4, 0x00 }, + { ADAU1373_DRC(0) + 0x5, 0xc0 }, + { ADAU1373_DRC(0) + 0x6, 0x00 }, + { ADAU1373_DRC(0) + 0x7, 0x00 }, + { ADAU1373_DRC(0) + 0x8, 0x00 }, + { ADAU1373_DRC(0) + 0x9, 0xc0 }, + { ADAU1373_DRC(0) + 0xa, 0x88 }, + { ADAU1373_DRC(0) + 0xb, 0x7a }, + { ADAU1373_DRC(0) + 0xc, 0xdf }, + { ADAU1373_DRC(0) + 0xd, 0x20 }, + { ADAU1373_DRC(0) + 0xe, 0x00 }, + { ADAU1373_DRC(0) + 0xf, 0x00 }, + { ADAU1373_DRC(1) + 0x0, 0x78 }, + { ADAU1373_DRC(1) + 0x1, 0x18 }, + { ADAU1373_DRC(1) + 0x2, 0x00 }, + { ADAU1373_DRC(1) + 0x3, 0x00 }, + { ADAU1373_DRC(1) + 0x4, 0x00 }, + { ADAU1373_DRC(1) + 0x5, 0xc0 }, + { ADAU1373_DRC(1) + 0x6, 0x00 }, + { ADAU1373_DRC(1) + 0x7, 0x00 }, + { ADAU1373_DRC(1) + 0x8, 0x00 }, + { ADAU1373_DRC(1) + 0x9, 0xc0 }, + { ADAU1373_DRC(1) + 0xa, 0x88 }, + { ADAU1373_DRC(1) + 0xb, 0x7a }, + { ADAU1373_DRC(1) + 0xc, 0xdf }, + { ADAU1373_DRC(1) + 0xd, 0x20 }, + { ADAU1373_DRC(1) + 0xe, 0x00 }, + { ADAU1373_DRC(1) + 0xf, 0x00 }, + { ADAU1373_DRC(2) + 0x0, 0x78 }, + { ADAU1373_DRC(2) + 0x1, 0x18 }, + { ADAU1373_DRC(2) + 0x2, 0x00 }, + { ADAU1373_DRC(2) + 0x3, 0x00 }, + { ADAU1373_DRC(2) + 0x4, 0x00 }, + { ADAU1373_DRC(2) + 0x5, 0xc0 }, + { ADAU1373_DRC(2) + 0x6, 0x00 }, + { ADAU1373_DRC(2) + 0x7, 0x00 }, + { ADAU1373_DRC(2) + 0x8, 0x00 }, + { ADAU1373_DRC(2) + 0x9, 0xc0 }, + { ADAU1373_DRC(2) + 0xa, 0x88 }, + { ADAU1373_DRC(2) + 0xb, 0x7a }, + { ADAU1373_DRC(2) + 0xc, 0xdf }, + { ADAU1373_DRC(2) + 0xd, 0x20 }, + { ADAU1373_DRC(2) + 0xe, 0x00 }, + { ADAU1373_DRC(2) + 0xf, 0x00 }, + { ADAU1373_3D_CTRL1, 0x00 }, + { ADAU1373_3D_CTRL2, 0x00 }, + { ADAU1373_FDSP_SEL1, 0x00 }, + { ADAU1373_FDSP_SEL2, 0x00 }, + { ADAU1373_FDSP_SEL2, 0x00 }, + { ADAU1373_FDSP_SEL4, 0x00 }, + { ADAU1373_DIGMICCTRL, 0x00 }, + { ADAU1373_DIGEN, 0x00 }, }; static const unsigned int adau1373_out_tlv[] = { @@ -418,6 +554,7 @@ static int adau1373_pll_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); unsigned int pll_id = w->name[3] - '1'; unsigned int val; @@ -426,7 +563,7 @@ static int adau1373_pll_event(struct snd_soc_dapm_widget *w, else val = 0; - snd_soc_update_bits(codec, ADAU1373_PLL_CTRL6(pll_id), + regmap_update_bits(adau1373->regmap, ADAU1373_PLL_CTRL6(pll_id), ADAU1373_PLL_CTRL6_PLL_EN, val); if (SND_SOC_DAPM_EVENT_ON(event)) @@ -938,7 +1075,7 @@ static int adau1373_hw_params(struct snd_pcm_substream *substream, adau1373_dai->enable_src = (div != 0); - snd_soc_update_bits(codec, ADAU1373_BCLKDIV(dai->id), + regmap_update_bits(adau1373->regmap, ADAU1373_BCLKDIV(dai->id), ADAU1373_BCLKDIV_SR_MASK | ADAU1373_BCLKDIV_BCLK_MASK, (div << 2) | ADAU1373_BCLKDIV_64); @@ -959,7 +1096,7 @@ static int adau1373_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - return snd_soc_update_bits(codec, ADAU1373_DAI(dai->id), + return regmap_update_bits(adau1373->regmap, ADAU1373_DAI(dai->id), ADAU1373_DAI_WLEN_MASK, ctrl); } @@ -1016,7 +1153,7 @@ static int adau1373_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - snd_soc_update_bits(codec, ADAU1373_DAI(dai->id), + regmap_update_bits(adau1373->regmap, ADAU1373_DAI(dai->id), ~ADAU1373_DAI_WLEN_MASK, ctrl); return 0; @@ -1039,7 +1176,7 @@ static int adau1373_set_dai_sysclk(struct snd_soc_dai *dai, adau1373_dai->sysclk = freq; adau1373_dai->clk_src = clk_id; - snd_soc_update_bits(dai->codec, ADAU1373_BCLKDIV(dai->id), + regmap_update_bits(adau1373->regmap, ADAU1373_BCLKDIV(dai->id), ADAU1373_BCLKDIV_SOURCE, clk_id << 5); return 0; @@ -1120,6 +1257,7 @@ static struct snd_soc_dai_driver adau1373_dai_driver[] = { static int adau1373_set_pll(struct snd_soc_codec *codec, int pll_id, int source, unsigned int freq_in, unsigned int freq_out) { + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); unsigned int dpll_div = 0; unsigned int x, r, n, m, i, j, mode; @@ -1187,36 +1325,36 @@ static int adau1373_set_pll(struct snd_soc_codec *codec, int pll_id, if (dpll_div) { dpll_div = 11 - dpll_div; - snd_soc_update_bits(codec, ADAU1373_PLL_CTRL6(pll_id), + regmap_update_bits(adau1373->regmap, ADAU1373_PLL_CTRL6(pll_id), ADAU1373_PLL_CTRL6_DPLL_BYPASS, 0); } else { - snd_soc_update_bits(codec, ADAU1373_PLL_CTRL6(pll_id), + regmap_update_bits(adau1373->regmap, ADAU1373_PLL_CTRL6(pll_id), ADAU1373_PLL_CTRL6_DPLL_BYPASS, ADAU1373_PLL_CTRL6_DPLL_BYPASS); } - snd_soc_write(codec, ADAU1373_DPLL_CTRL(pll_id), + regmap_write(adau1373->regmap, ADAU1373_DPLL_CTRL(pll_id), (source << 4) | dpll_div); - snd_soc_write(codec, ADAU1373_PLL_CTRL1(pll_id), (m >> 8) & 0xff); - snd_soc_write(codec, ADAU1373_PLL_CTRL2(pll_id), m & 0xff); - snd_soc_write(codec, ADAU1373_PLL_CTRL3(pll_id), (n >> 8) & 0xff); - snd_soc_write(codec, ADAU1373_PLL_CTRL4(pll_id), n & 0xff); - snd_soc_write(codec, ADAU1373_PLL_CTRL5(pll_id), + regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL1(pll_id), (m >> 8) & 0xff); + regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL2(pll_id), m & 0xff); + regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL3(pll_id), (n >> 8) & 0xff); + regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL4(pll_id), n & 0xff); + regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL5(pll_id), (r << 3) | (x << 1) | mode); /* Set sysclk to pll_rate / 4 */ - snd_soc_update_bits(codec, ADAU1373_CLK_SRC_DIV(pll_id), 0x3f, 0x09); + regmap_update_bits(adau1373->regmap, ADAU1373_CLK_SRC_DIV(pll_id), 0x3f, 0x09); return 0; } -static void adau1373_load_drc_settings(struct snd_soc_codec *codec, +static void adau1373_load_drc_settings(struct adau1373 *adau1373, unsigned int nr, uint8_t *drc) { unsigned int i; for (i = 0; i < ADAU1373_DRC_SIZE; ++i) - snd_soc_write(codec, ADAU1373_DRC(nr) + i, drc[i]); + regmap_write(adau1373->regmap, ADAU1373_DRC(nr) + i, drc[i]); } static bool adau1373_valid_micbias(enum adau1373_micbias_voltage micbias) @@ -1235,13 +1373,14 @@ static bool adau1373_valid_micbias(enum adau1373_micbias_voltage micbias) static int adau1373_probe(struct snd_soc_codec *codec) { + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); struct adau1373_platform_data *pdata = codec->dev->platform_data; bool lineout_differential = false; unsigned int val; int ret; int i; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); if (ret) { dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); return ret; @@ -1256,7 +1395,7 @@ static int adau1373_probe(struct snd_soc_codec *codec) return -EINVAL; for (i = 0; i < pdata->num_drc; ++i) { - adau1373_load_drc_settings(codec, i, + adau1373_load_drc_settings(adau1373, i, pdata->drc_setting[i]); } @@ -1268,18 +1407,18 @@ static int adau1373_probe(struct snd_soc_codec *codec) if (pdata->input_differential[i]) val |= BIT(i); } - snd_soc_write(codec, ADAU1373_INPUT_MODE, val); + regmap_write(adau1373->regmap, ADAU1373_INPUT_MODE, val); val = 0; if (pdata->lineout_differential) val |= ADAU1373_OUTPUT_CTRL_LDIFF; if (pdata->lineout_ground_sense) val |= ADAU1373_OUTPUT_CTRL_LNFBEN; - snd_soc_write(codec, ADAU1373_OUTPUT_CTRL, val); + regmap_write(adau1373->regmap, ADAU1373_OUTPUT_CTRL, val); lineout_differential = pdata->lineout_differential; - snd_soc_write(codec, ADAU1373_EP_CTRL, + regmap_write(adau1373->regmap, ADAU1373_EP_CTRL, (pdata->micbias1 << ADAU1373_EP_CTRL_MICBIAS1_OFFSET) | (pdata->micbias2 << ADAU1373_EP_CTRL_MICBIAS2_OFFSET)); } @@ -1289,7 +1428,7 @@ static int adau1373_probe(struct snd_soc_codec *codec) ARRAY_SIZE(adau1373_lineout2_controls)); } - snd_soc_write(codec, ADAU1373_ADC_CTRL, + regmap_write(adau1373->regmap, ADAU1373_ADC_CTRL, ADAU1373_ADC_CTRL_RESET_FORCE | ADAU1373_ADC_CTRL_PEAK_DETECT); return 0; @@ -1298,17 +1437,19 @@ static int adau1373_probe(struct snd_soc_codec *codec) static int adau1373_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); + switch (level) { case SND_SOC_BIAS_ON: break; case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - snd_soc_update_bits(codec, ADAU1373_PWDN_CTRL3, + regmap_update_bits(adau1373->regmap, ADAU1373_PWDN_CTRL3, ADAU1373_PWDN_CTRL3_PWR_EN, ADAU1373_PWDN_CTRL3_PWR_EN); break; case SND_SOC_BIAS_OFF: - snd_soc_update_bits(codec, ADAU1373_PWDN_CTRL3, + regmap_update_bits(adau1373->regmap, ADAU1373_PWDN_CTRL3, ADAU1373_PWDN_CTRL3_PWR_EN, 0); break; } @@ -1324,17 +1465,49 @@ static int adau1373_remove(struct snd_soc_codec *codec) static int adau1373_suspend(struct snd_soc_codec *codec) { - return adau1373_set_bias_level(codec, SND_SOC_BIAS_OFF); + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = adau1373_set_bias_level(codec, SND_SOC_BIAS_OFF); + regcache_cache_only(adau1373->regmap, true); + + return ret; } static int adau1373_resume(struct snd_soc_codec *codec) { + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(adau1373->regmap, false); adau1373_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_cache_sync(codec); + regcache_sync(adau1373->regmap); return 0; } +static bool adau1373_register_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case ADAU1373_SOFT_RESET: + case ADAU1373_ADC_DAC_STATUS: + return true; + default: + return false; + } +} + +static const struct regmap_config adau1373_regmap_config = { + .val_bits = 8, + .reg_bits = 8, + + .volatile_reg = adau1373_register_volatile, + .max_register = ADAU1373_SOFT_RESET, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = adau1373_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(adau1373_reg_defaults), +}; + static struct snd_soc_codec_driver adau1373_codec_driver = { .probe = adau1373_probe, .remove = adau1373_remove, @@ -1342,9 +1515,6 @@ static struct snd_soc_codec_driver adau1373_codec_driver = { .resume = adau1373_resume, .set_bias_level = adau1373_set_bias_level, .idle_bias_off = true, - .reg_cache_size = ARRAY_SIZE(adau1373_default_regs), - .reg_cache_default = adau1373_default_regs, - .reg_word_size = sizeof(uint8_t), .set_pll = adau1373_set_pll, @@ -1366,6 +1536,11 @@ static int adau1373_i2c_probe(struct i2c_client *client, if (!adau1373) return -ENOMEM; + adau1373->regmap = devm_regmap_init_i2c(client, + &adau1373_regmap_config); + if (IS_ERR(adau1373->regmap)) + return PTR_ERR(adau1373->regmap); + dev_set_drvdata(&client->dev, adau1373); ret = snd_soc_register_codec(&client->dev, &adau1373_codec_driver, -- 1.8.0

2 3

[alsa-devel] [RFC/RFT PATCH] ALSA: hda - hdmi: Add ATI/AMD multi-channel audio support
by Anssi Hannula 27 Sep '13

27 Sep '13

Request for comments and testing. AMD just released the documentation needed for multi-channel HDMI support, so here is the initial patch adding such support. This has been tested on codec 0x1002aa01, revision 0x100200 with the exception of custom channel maps, which I intend to test later. I'm especially interested in testers with: - Older codecs other than 0x1002aa01. My best guess is that the new code should work on them as well, but I'm not really 100% sure. - Codec 0x1002aa01 revisions 0x100300 (Rev ID 3) or later (see /proc/asound/cardX/codec#Y). These support individual channel remapping and that code is entirely untested (even for regular 5.1/7.1). I decided it was best to get this out there early even though major parts of it are still untested, for comments and for other people to test :) Note that this recent commit is probably required for multi-channel to work with the open-source radeon driver: ALSA: hda - hdmi: Fallback to ALSA allocation when selecting CA https://git.kernel.org/cgit/linux/kernel/git/torvalds/linux.git/commit/soun… Easiest way to test basic multichannel is by speaker-test, e.g. speaker-test -D hdmi:CARD=Generic,DEV=0 -c8 -r192000 -F S16_LE for 7.1 16-bit 192kHz Detailed discussion below: ATI/AMD codecs do not support all the standard HDA HDMI/DP functions. In particular, these are missing: - ELD support - standard channel mapping support - standard infoframe configuration support In place of them, custom HDA pin verbs are included, that allow: - getting speaker mask from EDID - getting SADs from EDID - getting connection type from EDID (HDMI, DisplayPort) - setting CA for infoframe - setting down-mix information for infoframe - channel pair remapping - individual channel remapping (revision ID 3+) The documentation for the verbs has now been released by AMD: http://www.x.org/docs/AMD/AMD_HDA_verbs.pdf Also, converted nodes do not have all their supported rates, formats and channel counts listed. Add support for the ATI/AMD specific verbs and use them instead of the generic methods on ATI/AMD codecs. This allows multi-channel audio to work. Is the way it is done OK, or should something more generic be put into place (custom callbacks and/or flags in struct hdmi_spec maybe)? On the older ATI/AMD codecs that do not support individual remapping, channel remapping is restricted to pair remapping. This does not affect the standard multi-channel configurations. Since there is no ELD support (unless out-of-band communication with the graphics driver is added so that they can provide it to us directly), ELD is currently emulated with only the basic information populated (speaker mask, SADs, connection type). There are some bits missing on the radeon driver side (the SADs etc are not populated) so currently the ELD emulation only works on fglrx. ELD emulation is not needed for multi-channel playback. Rafał, are you interested in implementing the missing bits in the radeon driver, as I see you have looked into this before? The ELD emulation was quite quickly hacked together and hasn't been cleaned up yet, since I'm not sure how we want to ultimately handle this. Is ELD emulation the way to go? If so, where does the code belong (the ATI specific code is in patch_hdmi.c, but the ELD parsing/defines etc are in hda_eld.c)? Codec 0x1002aa01 is switched back to patch_atihdmi to support the ATI/AMD specific features. HBR mode (needed for DTS-HD/TrueHD passthrough) is not working yet, and the released documentation does not mention it. Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> Tested-by: Peter Frühberger <fritsch(a)xbmc.org> Cc: Rafał Miłecki <zajec5(a)gmail.com> --- sound/pci/hda/patch_hdmi.c | 418 ++++++++++++++++++++++++++++++++++++++++----- 1 file changed, 374 insertions(+), 44 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 3d8cd044..741d75e 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -6,6 +6,7 @@ * Copyright (c) 2006 ATI Technologies Inc. * Copyright (c) 2008 NVIDIA Corp. All rights reserved. * Copyright (c) 2008 Wei Ni <wni(a)nvidia.com> + * Copyright (c) 2013 Anssi Hannula <anssi.hannula(a)iki.fi> * * Authors: * Wu Fengguang <wfg(a)linux.intel.com> @@ -45,6 +46,9 @@ module_param(static_hdmi_pcm, bool, 0644); MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info"); #define is_haswell(codec) ((codec)->vendor_id == 0x80862807) +#define is_atihdmi(codec) (((codec)->vendor_id & 0xffff0000) == 0x10020000) +#define has_amd_full_remap_support(codec) \ + ((codec)->vendor_id == 0x1002791a && ((codec)->revision_id & 0xff00) >= 0x0300) struct hdmi_spec_per_cvt { hda_nid_t cvt_nid; @@ -89,7 +93,7 @@ struct hdmi_spec { struct hdmi_eld temp_eld; /* - * Non-generic ATI/NVIDIA specific + * Non-generic VIA/NVIDIA specific */ struct hda_multi_out multiout; struct hda_pcm_stream pcm_playback; @@ -573,6 +577,20 @@ static int hdmi_channel_allocation(struct hdmi_eld *eld, int channels) return ca; } +#ifdef CONFIG_SND_DEBUG_VERBOSE +static int atihdmi_get_chan_slot(struct hda_codec *codec, hda_nid_t pin_nid, int asp_slot); + +static int hdmi_get_chan_slot(struct hda_codec *codec, hda_nid_t pin_nid, int asp_slot) +{ + if (is_atihdmi(codec)) + return atihdmi_get_chan_slot(codec, pin_nid, asp_slot); + + return snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_HDMI_CHAN_SLOT, + asp_slot); +} +#endif + static void hdmi_debug_channel_mapping(struct hda_codec *codec, hda_nid_t pin_nid) { @@ -581,14 +599,26 @@ static void hdmi_debug_channel_mapping(struct hda_codec *codec, int slot; for (i = 0; i < 8; i++) { - slot = snd_hda_codec_read(codec, pin_nid, 0, - AC_VERB_GET_HDMI_CHAN_SLOT, i); + slot = hdmi_get_chan_slot(codec, pin_nid, i); printk(KERN_DEBUG "HDMI: ASP channel %d => slot %d\n", slot >> 4, slot & 0xf); } #endif } +static int atihdmi_set_chan_slot(struct hda_codec *codec, hda_nid_t pin_nid, + int chanslot_setup); + +static int hdmi_set_chan_slot(struct hda_codec *codec, hda_nid_t pin_nid, + int chanslot_setup) +{ + if (is_atihdmi(codec)) + return atihdmi_set_chan_slot(codec, pin_nid, chanslot_setup); + + return snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_HDMI_CHAN_SLOT, + chanslot_setup); +} static void hdmi_std_setup_channel_mapping(struct hda_codec *codec, hda_nid_t pin_nid, @@ -617,9 +647,8 @@ static void hdmi_std_setup_channel_mapping(struct hda_codec *codec, } for (i = 0; i < 8; i++) { - err = snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_HDMI_CHAN_SLOT, - non_pcm ? non_pcm_mapping[i] : hdmi_channel_mapping[ca][i]); + err = hdmi_set_chan_slot(codec, pin_nid, + non_pcm ? non_pcm_mapping[i] : hdmi_channel_mapping[ca][i]); if (err) { snd_printdd(KERN_NOTICE "HDMI: channel mapping failed\n"); @@ -728,8 +757,7 @@ static int hdmi_manual_setup_channel_mapping(struct hda_codec *codec, else val = 0xf; val |= (i << 4); - err = snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_HDMI_CHAN_SLOT, val); + err = hdmi_set_chan_slot(codec, pin_nid, val); if (err) return -EINVAL; } @@ -883,6 +911,8 @@ static bool hdmi_infoframe_uptodate(struct hda_codec *codec, hda_nid_t pin_nid, return true; } +static void atihdmi_set_ca(struct hda_codec *codec, hda_nid_t pin_nid, int ca); + static void hdmi_setup_audio_infoframe(struct hda_codec *codec, struct hdmi_spec_per_pin *per_pin, bool non_pcm) @@ -912,6 +942,16 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, if (ca < 0) ca = 0; + if (is_atihdmi(codec)) { + /* for ATI/AMD we just want to map channels and set ca */ + hdmi_setup_channel_mapping(codec, pin_nid, non_pcm, ca, + channels, per_pin->chmap, + per_pin->chmap_set); + atihdmi_set_ca(codec, pin_nid, ca); + per_pin->non_pcm = non_pcm; + return; + } + memset(&ai, 0, sizeof(ai)); if (eld->info.conn_type == 0) { /* HDMI */ struct hdmi_audio_infoframe *hdmi_ai = &ai.hdmi; @@ -1274,6 +1314,9 @@ static int hdmi_read_pin_conn(struct hda_codec *codec, int pin_idx) return 0; } +static int atihdmi_get_eld(struct hda_codec *codec, hda_nid_t nid, + unsigned char *buf, int *eld_size); + static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) { struct hda_codec *codec = per_pin->codec; @@ -1303,8 +1346,12 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) "HDMI status: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n", codec->addr, pin_nid, pin_eld->monitor_present, eld->eld_valid); + if (pin_eld->monitor_present && is_atihdmi(codec)) + eld->eld_valid = (atihdmi_get_eld(codec, pin_nid, eld->eld_buffer, &eld->eld_size) == 0); + if (eld->eld_valid) { - if (snd_hdmi_get_eld(codec, pin_nid, eld->eld_buffer, + if (!is_atihdmi(codec) && + snd_hdmi_get_eld(codec, pin_nid, eld->eld_buffer, &eld->eld_size) < 0) eld->eld_valid = false; else { @@ -1603,6 +1650,8 @@ static int hdmi_chmap_ctl_info(struct snd_kcontrol *kcontrol, return 0; } +static int atihdmi_swap_fc_lfe(int pos); + static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, unsigned int size, unsigned int __user *tlv) { @@ -1613,6 +1662,10 @@ static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, FL | FR | RL | RR | LFE | FC | RLC | RRC; unsigned int __user *dst; int chs, count = 0; + int tlv_type = SNDRV_CTL_TLVT_CHMAP_VAR; + + if (is_atihdmi(codec) && !has_amd_full_remap_support(codec)) + tlv_type = SNDRV_CTL_TLVT_CHMAP_PAIRED; if (size < 8) return -ENOMEM; @@ -1620,19 +1673,33 @@ static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, return -EFAULT; size -= 8; dst = tlv + 2; - for (chs = 2; chs <= spec->channels_max; chs++) { + for (chs = 2; chs <= spec->channels_max; + chs += (tlv_type == SNDRV_CTL_TLVT_CHMAP_PAIRED) ? 2 : 1) { int i, c; struct cea_channel_speaker_allocation *cap; cap = channel_allocations; for (i = 0; i < ARRAY_SIZE(channel_allocations); i++, cap++) { int chs_bytes = chs * 4; - if (cap->channels != chs) + if (tlv_type == SNDRV_CTL_TLVT_CHMAP_PAIRED) { + int chanpairs = 0; + /* in paired mode we need to take into account + * the occupied channel pairs instead of just the + * channel count */ + for (c = 0; c < 8; c++) { + if (cap->speakers[c] || cap->speakers[c+1]) + chanpairs++; + } + if (chanpairs * 2 != chs) + continue; + + } else if (cap->channels != chs) continue; + if (cap->spk_mask & ~valid_mask) continue; if (size < 8) return -ENOMEM; - if (put_user(SNDRV_CTL_TLVT_CHMAP_VAR, dst) || + if (put_user(tlv_type, dst) || put_user(chs_bytes, dst + 1)) return -EFAULT; dst += 2; @@ -1643,10 +1710,25 @@ static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, size -= chs_bytes; count += chs_bytes; for (c = 7; c >= 0; c--) { - int spk = cap->speakers[c]; - if (!spk) - continue; - if (put_user(spk_to_chmap(spk), dst)) + int spk; + int chan = c; + int chpos; + + if (tlv_type == SNDRV_CTL_TLVT_CHMAP_PAIRED) + chan = 7 - atihdmi_swap_fc_lfe(7 - chan); + + spk = cap->speakers[chan]; + if (spk) + chpos = spk_to_chmap(spk); + else { + /* We need to reserve an N/A channel in paired mode */ + if (tlv_type == SNDRV_CTL_TLVT_CHMAP_PAIRED) + chpos = SNDRV_CHMAP_NA; + else + continue; + } + + if (put_user(chpos, dst)) return -EFAULT; dst++; } @@ -1672,6 +1754,18 @@ static int hdmi_chmap_ctl_get(struct snd_kcontrol *kcontrol, return 0; } +static int atihdmi_pairwise_chmap_check_order(struct hda_codec *codec, int ca, + int chs, unsigned char *map); + +static int hdmi_chmap_check_order(struct hda_codec *codec, int ca, + int chs, unsigned char *map) +{ + if (is_atihdmi(codec) && !has_amd_full_remap_support(codec)) + return atihdmi_pairwise_chmap_check_order(codec, ca, chs, map); + + return 0; /* anything can be remapped as needed */ +} + static int hdmi_chmap_ctl_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1683,7 +1777,7 @@ static int hdmi_chmap_ctl_put(struct snd_kcontrol *kcontrol, unsigned int ctl_idx; struct snd_pcm_substream *substream; unsigned char chmap[8]; - int i, ca, prepared = 0; + int i, err, ca, prepared = 0; ctl_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); substream = snd_pcm_chmap_substream(info, ctl_idx); @@ -1707,6 +1801,9 @@ static int hdmi_chmap_ctl_put(struct snd_kcontrol *kcontrol, ca = hdmi_manual_channel_allocation(ARRAY_SIZE(chmap), chmap); if (ca < 0) return -EINVAL; + err = hdmi_chmap_check_order(codec, ca, ARRAY_SIZE(chmap), chmap); + if (err < 0) + return -EINVAL; per_pin->chmap_set = true; memcpy(per_pin->chmap, chmap, sizeof(chmap)); if (prepared) @@ -2551,48 +2648,281 @@ static int patch_nvhdmi_8ch_7x(struct hda_codec *codec) /* * ATI-specific implementations - * - * FIXME: we may omit the whole this and use the generic code once after - * it's confirmed to work. */ -#define ATIHDMI_CVT_NID 0x02 /* audio converter */ -#define ATIHDMI_PIN_NID 0x03 /* HDMI output pin */ +/* ATI/AMD specific HDA pin verbs, see the AMD HDA Verbs specification */ +#define ATI_VERB_SET_CHANNEL_ALLOCATION 0x771 +#define ATI_VERB_SET_DOWNMIX_INFO 0x772 +#define ATI_VERB_SET_AUDIO_DESCRIPTOR 0x776 +#define ATI_VERB_SET_MULTICHANNEL_01 0x777 +#define ATI_VERB_SET_MULTICHANNEL_23 0x778 +#define ATI_VERB_SET_MULTICHANNEL_45 0x779 +#define ATI_VERB_SET_MULTICHANNEL_67 0x77a +#define ATI_VERB_SET_MULTICHANNEL_1 0x785 +#define ATI_VERB_SET_MULTICHANNEL_3 0x786 +#define ATI_VERB_SET_MULTICHANNEL_5 0x787 +#define ATI_VERB_SET_MULTICHANNEL_7 0x788 +#define ATI_VERB_SET_MULTICHANNEL_MODE 0x789 +#define ATI_VERB_GET_SPEAKER_ALLOCATION 0xf70 +#define ATI_VERB_GET_CHANNEL_ALLOCATION 0xf71 +#define ATI_VERB_GET_DOWNMIX_INFO 0xf72 +#define ATI_VERB_GET_AUDIO_DESCRIPTOR 0xf76 +#define ATI_VERB_GET_MULTICHANNEL_01 0xf77 +#define ATI_VERB_GET_MULTICHANNEL_23 0xf78 +#define ATI_VERB_GET_MULTICHANNEL_45 0xf79 +#define ATI_VERB_GET_MULTICHANNEL_67 0xf7a +#define ATI_VERB_GET_MULTICHANNEL_1 0xf85 +#define ATI_VERB_GET_MULTICHANNEL_3 0xf86 +#define ATI_VERB_GET_MULTICHANNEL_5 0xf87 +#define ATI_VERB_GET_MULTICHANNEL_7 0xf88 +#define ATI_VERB_GET_MULTICHANNEL_MODE 0xf89 + +#define ATI_OUT_ENABLE 0x1 + +/* TODO: cleanup this function */ +static int atihdmi_get_eld(struct hda_codec *codec, hda_nid_t nid, + unsigned char *buf, int *eld_size) +{ + int spkalloc, ati_sad; + int pos, i; + + /* ATI/AMD does not have ELD, emulate it */ + + spkalloc = snd_hda_codec_read(codec, nid, 0, ATI_VERB_GET_SPEAKER_ALLOCATION, 0); + + if (!spkalloc) + return -EINVAL; + + memset(buf, 0, 84); + + /* speaker allocation from EDID */ + buf[7] = spkalloc & 0x00007f; + + /* is DisplayPort? */ + if (spkalloc & 0x000200) + buf[5] |= 0x04; + + pos = 20; + for (i = 1; i <= 14; i++) { + if (i == 13) /* not handled by ATI/AMD */ + continue; + + snd_hda_codec_write(codec, nid, 0, ATI_VERB_SET_AUDIO_DESCRIPTOR, i << 3); + ati_sad = snd_hda_codec_read(codec, nid, 0, ATI_VERB_GET_AUDIO_DESCRIPTOR, 0); + + if (ati_sad & 0xff00) { + memcpy(buf + pos, &ati_sad, 3); + pos += 3; + } + + if (i == 1 && ati_sad & 0xff000000 && (ati_sad & 0xff00) != (ati_sad & 0xff000000) >> 16) { + /* for PCM there is a separate stereo rate mask */ + memcpy(buf + pos, &ati_sad, 3); + /* max channels to 2 */ + buf[pos+0] = (buf[pos+0] & ~0x7) | 0x1; + /* rates from extra byte */ + buf[pos+1] = (ati_sad & 0xff000000) >> 24; + pos += 3; + } + } + + if (pos == 20) + return -EINVAL; + + /* version */ + buf[0] = 0x10; + + /* Baseline length */ + buf[2] = pos - 4; + + /* SAD count */ + buf[5] |= ((pos - 20) / 3) << 4; + + *eld_size = pos; + + return 0; +} + +static void atihdmi_set_ca(struct hda_codec *codec, hda_nid_t pin_nid, int ca) +{ + printk("ATI: setting ca %d\n", ca); + snd_hda_codec_write(codec, pin_nid, 0, ATI_VERB_SET_CHANNEL_ALLOCATION, ca); +} + +static int atihdmi_swap_fc_lfe(int pos) +{ + /* + * Older ATI/AMD without channel-wise mapping + * have automatic FC/LFE swap built-in. + */ + + switch (pos) { + /* see channel_allocations[].speakers[] */ + case 2: return 3; + case 3: return 2; + default: break; + } + + return pos; +} + +static int atihdmi_pairwise_chmap_check_order(struct hda_codec *codec, int ca, + int chs, unsigned char *map) +{ + struct cea_channel_speaker_allocation *cap; + int i, j; + + /* check that only channel pairs need to be remapped on old ATI/AMD */ + + cap = &channel_allocations[get_channel_allocation_order(ca)]; + for (i = 0; i < chs; ++i) { + int mask = to_spk_mask(map[i]); + bool ok = false; + bool companion_ok = false; + + if (!mask) + continue; + + for (j = 0 + i % 2; j < 8; j += 2) { + int chan_idx = 7 - atihdmi_swap_fc_lfe(j); + if (cap->speakers[chan_idx] == mask) { + /* channel is in a supported position */ + ok = true; + + if (i % 2 == 0 && i + 1 < chs) { + /* even channel, check the odd companion */ + int comp_chan_idx = 7 - atihdmi_swap_fc_lfe(j + 1); + int comp_mask_req = to_spk_mask(map[i+1]); + int comp_mask_act = cap->speakers[comp_chan_idx]; + + if (comp_mask_req == comp_mask_act) + companion_ok = true; + else + return -EINVAL; + } + break; + } + } + + if (!ok) + return -EINVAL; + + if (companion_ok) + i++; /* companion channel already checked */ + } + + return 0; +} + +static int atihdmi_set_chan_slot(struct hda_codec *codec, hda_nid_t pin_nid, + int chanslot_setup) +{ + int hdmi_slot = chanslot_setup & 0xf; + int stream_channel = chanslot_setup >> 4; + int verb; + int ati_channel_setup = 0; + + if (!has_amd_full_remap_support(codec)) { + hdmi_slot = atihdmi_swap_fc_lfe(hdmi_slot); + + /* In case this is an odd slot but without stream channel, do not + * disable the slot since the corresponding even slot could have a + * channel. In case neither have a channel, the slot pair will be + * disabled when this function is called for the even slot. */ + if (hdmi_slot % 2 != 0 && stream_channel == 0xf) + return 0; + + hdmi_slot -= hdmi_slot % 2; + + if (stream_channel != 0xf) + stream_channel -= stream_channel % 2; + } + + verb = ATI_VERB_SET_MULTICHANNEL_01 + hdmi_slot/2 + (hdmi_slot % 2) * 0x00e; + + /* ati_channel_setup format: [7..4] = stream_channel_id, [1] = mute, [0] = enable */ + + if (stream_channel != 0xf) + ati_channel_setup = (stream_channel << 4) | ATI_OUT_ENABLE; + + return snd_hda_codec_write(codec, pin_nid, 0, verb, ati_channel_setup); +} + +#ifdef CONFIG_SND_DEBUG_VERBOSE +static int atihdmi_get_chan_slot(struct hda_codec *codec, hda_nid_t pin_nid, int asp_slot) +{ + bool was_odd = false; + int ati_asp_slot = asp_slot; + int verb; + int ati_channel_setup; + + /* emulate AC_VERB_GET_HDMI_CHAN_SLOT */ + + if (!has_amd_full_remap_support(codec)) { + ati_asp_slot = atihdmi_swap_fc_lfe(asp_slot); + if (ati_asp_slot % 2 != 0) { + ati_asp_slot -= 1; + was_odd = true; + } + } -static int atihdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) + verb = ATI_VERB_GET_MULTICHANNEL_01 + ati_asp_slot/2 + (ati_asp_slot % 2) * 0x00e; + + ati_channel_setup = snd_hda_codec_read(codec, pin_nid, 0, verb, 0); + + if (!(ati_channel_setup & ATI_OUT_ENABLE)) + return (0xf << 4) | asp_slot; + + return ((ati_channel_setup & 0xf0) + ((!!was_odd) << 4)) | asp_slot; +} +#endif + +static int atihdmi_init(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; - struct hdmi_spec_per_cvt *per_cvt = get_cvt(spec, 0); - int chans = substream->runtime->channels; - int i, err; + int pin_idx, err; - err = simple_playback_pcm_prepare(hinfo, codec, stream_tag, format, - substream); - if (err < 0) + err = generic_hdmi_init(codec); + + if (err) return err; - snd_hda_codec_write(codec, per_cvt->cvt_nid, 0, - AC_VERB_SET_CVT_CHAN_COUNT, chans - 1); - /* FIXME: XXX */ - for (i = 0; i < chans; i++) { - snd_hda_codec_write(codec, per_cvt->cvt_nid, 0, - AC_VERB_SET_HDMI_CHAN_SLOT, - (i << 4) | i); + + for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { + struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); + + snd_hda_codec_write(codec, per_pin->pin_nid, 0, ATI_VERB_SET_DOWNMIX_INFO, 0); + + /* enable channel-wise remap mode if supported */ + if (has_amd_full_remap_support(codec)) + snd_hda_codec_write(codec, per_pin->pin_nid, 0, ATI_VERB_SET_MULTICHANNEL_MODE, 1); } + return 0; } static int patch_atihdmi(struct hda_codec *codec) { struct hdmi_spec *spec; - int err = patch_simple_hdmi(codec, ATIHDMI_CVT_NID, ATIHDMI_PIN_NID); - if (err < 0) + struct hdmi_spec_per_cvt *per_cvt; + int err, cvt_idx; + + err = patch_generic_hdmi(codec); + + if (err) return err; + + codec->patch_ops.init = atihdmi_init; + + /* ATI/AMD converters do not advertise all of their capabilities */ spec = codec->spec; - spec->pcm_playback.ops.prepare = atihdmi_playback_pcm_prepare; + for (cvt_idx = 0; cvt_idx < spec->num_cvts; cvt_idx++) { + per_cvt = get_cvt(spec, cvt_idx); + per_cvt->channels_max = max(per_cvt->channels_max, 8u); + per_cvt->rates |= SUPPORTED_RATES; + per_cvt->formats |= SUPPORTED_FORMATS; + per_cvt->maxbps = max(per_cvt->maxbps, 24u); + } + return 0; } @@ -2612,7 +2942,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x1002793c, .name = "RS600 HDMI", .patch = patch_atihdmi }, { .id = 0x10027919, .name = "RS600 HDMI", .patch = patch_atihdmi }, { .id = 0x1002791a, .name = "RS690/780 HDMI", .patch = patch_atihdmi }, -{ .id = 0x1002aa01, .name = "R6xx HDMI", .patch = patch_generic_hdmi }, +{ .id = 0x1002aa01, .name = "R6xx HDMI", .patch = patch_atihdmi }, { .id = 0x10951390, .name = "SiI1390 HDMI", .patch = patch_generic_hdmi }, { .id = 0x10951392, .name = "SiI1392 HDMI", .patch = patch_generic_hdmi }, { .id = 0x17e80047, .name = "Chrontel HDMI", .patch = patch_generic_hdmi }, -- 1.8.1.5

5 10

[alsa-devel] pcm_rate API question
by Rob Sykes 27 Sep '13

27 Sep '13

Hello alsa devs, There's an ongoing attempt to see if libsoxr-lsr can be used as a plug-in replacement for libsamplerate in pulseaudio and in alsa (libsoxr-lsr is designed with this usage in mind). With pulseaudio, it seems to 'just work', but with alsa, it looks like there is a buffering problem. Progress reports for this are here: http://blog.ivitera.com/pavel/linux-audio/pulseaudio-with-ld_preloading-lib… http://blog.ivitera.com/pavel/linux-audio/simple-use-of-libsoxr-lsr-in-alsa… AFAICT, the problem with alsa relates to the use of this function in pcm_rate.h: typedef struct snd_pcm_rate_ops { ... void (*convert_s16)(void *obj, int16_t *dst, unsigned int dst_frames, const int16_t *src, unsigned int src_frames); Is there documentation available regarding the use of this function? For example, is the plugin obliged always to consume src_frames and generate dst_frames samples? If so, how is the plugin expected to comply (say, if it's not been given enough input to generate that amount of output)? TIA, Rob

2 1

[alsa-devel] [PATCH] ASoC: wm8993: drop regulator_bulk_free of devm_ allocated data
by Wei Yongjun 27 Sep '13

27 Sep '13

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> It's not necessary to free regulator consumers allocated with devm_regulator_bulk_get. Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/codecs/wm8993.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 433d59a..2ee23a3 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1562,7 +1562,6 @@ static int wm8993_remove(struct snd_soc_codec *codec) struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec); wm8993_set_bias_level(codec, SND_SOC_BIAS_OFF); - regulator_bulk_free(ARRAY_SIZE(wm8993->supplies), wm8993->supplies); return 0; }

2 1

[alsa-devel] [PATCH] ASoC: wm8993: drop regulator_bulk_free of devm_ allocated data
by Wei Yongjun 27 Sep '13

27 Sep '13

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> It's not necessary to free regulator consumers allocated with devm_regulator_bulk_get. Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/codecs/wm8993.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 433d59a..2ee23a3 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1562,7 +1562,6 @@ static int wm8993_remove(struct snd_soc_codec *codec) struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec); wm8993_set_bias_level(codec, SND_SOC_BIAS_OFF); - regulator_bulk_free(ARRAY_SIZE(wm8993->supplies), wm8993->supplies); return 0; }

1 0

[alsa-devel] [PATCH] ASoC: rcar: remove unnecessary mach/clock.h
by Kuninori Morimoto 27 Sep '13

27 Sep '13

${LINUX}/sound/soc/sh driver can be compiled from SuperH and ARM. but, ${LINUX}/sound/soc/sh/rcar driver included SH-ARM specific header. This patch removes it Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/adg.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index ad9ceb9..9430097 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -8,7 +8,6 @@ * for more details. */ #include <linux/sh_clk.h> -#include <mach/clock.h> #include "rsnd.h" #define CLKA 0 -- 1.7.9.5

3 2

[alsa-devel] [PATCH] ASoC: fsl/imx-sgtl5000: Fix uninitialized pointer use in error path
by Philipp Zabel 26 Sep '13

26 Sep '13

This patch avoids to dereference the uninitialized data pointer if the error path is entered before devm_kzalloc is called (or if the allocation fails). It fixes the following warning: sound/soc/fsl/imx-sgtl5000.c: In function 'imx_sgtl5000_probe': sound/soc/fsl/imx-sgtl5000.c:175:18: warning: 'data' may be used uninitialized in this function [-Wmaybe-uninitialized] Signed-off-by: Philipp Zabel <p.zabel(a)pengutronix.de> --- sound/soc/fsl/imx-sgtl5000.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index c49e4cb..6f4bdc8 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -62,7 +62,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) struct device_node *ssi_np, *codec_np; struct platform_device *ssi_pdev; struct i2c_client *codec_dev; - struct imx_sgtl5000_data *data; + struct imx_sgtl5000_data *data = NULL; int int_port, ext_port; int ret; @@ -172,7 +172,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) return 0; fail: - if (!IS_ERR(data->codec_clk)) + if (data && !IS_ERR(data->codec_clk)) clk_put(data->codec_clk); if (ssi_np) of_node_put(ssi_np); -- 1.8.4.rc3

2 1

Re: [alsa-devel] [PATCH 39/51] DMA-API: others: use dma_set_coherent_mask()
by Archit Taneja 26 Sep '13

26 Sep '13

Hi, On Friday 20 September 2013 04:41 AM, Russell King wrote: > The correct way for a driver to specify the coherent DMA mask is > not to directly access the field in the struct device, but to use > dma_set_coherent_mask(). Only arch and bus code should access this > member directly. > > Convert all direct write accesses to using the correct API. > > Signed-off-by: Russell King <rmk+kernel(a)arm.linux.org.uk> > --- > drivers/ata/pata_ixp4xx_cf.c | 5 ++++- > drivers/gpu/drm/exynos/exynos_drm_drv.c | 6 +++++- > drivers/gpu/drm/omapdrm/omap_dmm_tiler.c | 5 +++-- > 3 files changed, 12 insertions(+), 4 deletions(-) > > diff --git a/drivers/ata/pata_ixp4xx_cf.c b/drivers/ata/pata_ixp4xx_cf.c > index 1ec53f8..ddf470c 100644 > --- a/drivers/ata/pata_ixp4xx_cf.c > +++ b/drivers/ata/pata_ixp4xx_cf.c > @@ -144,6 +144,7 @@ static int ixp4xx_pata_probe(struct platform_device *pdev) > struct ata_host *host; > struct ata_port *ap; > struct ixp4xx_pata_data *data = dev_get_platdata(&pdev->dev); > + int ret; > > cs0 = platform_get_resource(pdev, IORESOURCE_MEM, 0); > cs1 = platform_get_resource(pdev, IORESOURCE_MEM, 1); > @@ -157,7 +158,9 @@ static int ixp4xx_pata_probe(struct platform_device *pdev) > return -ENOMEM; > > /* acquire resources and fill host */ > - pdev->dev.coherent_dma_mask = DMA_BIT_MASK(32); > + ret = dma_set_coherent_mask(&pdev->dev, DMA_BIT_MASK(32)); > + if (ret) > + return ret; > > data->cs0 = devm_ioremap(&pdev->dev, cs0->start, 0x1000); > data->cs1 = devm_ioremap(&pdev->dev, cs1->start, 0x1000); > diff --git a/drivers/gpu/drm/exynos/exynos_drm_drv.c b/drivers/gpu/drm/exynos/exynos_drm_drv.c > index bb82ef7..81192d0 100644 > --- a/drivers/gpu/drm/exynos/exynos_drm_drv.c > +++ b/drivers/gpu/drm/exynos/exynos_drm_drv.c > @@ -286,7 +286,11 @@ static struct drm_driver exynos_drm_driver = { > > static int exynos_drm_platform_probe(struct platform_device *pdev) > { > - pdev->dev.coherent_dma_mask = DMA_BIT_MASK(32); > + int ret; > + > + ret = dma_set_coherent_mask(&pdev->dev, DMA_BIT_MASK(32)); > + if (ret) > + return ret; > > return drm_platform_init(&exynos_drm_driver, pdev); > } > diff --git a/drivers/gpu/drm/omapdrm/omap_dmm_tiler.c b/drivers/gpu/drm/omapdrm/omap_dmm_tiler.c > index acf6678..701c4c1 100644 > --- a/drivers/gpu/drm/omapdrm/omap_dmm_tiler.c > +++ b/drivers/gpu/drm/omapdrm/omap_dmm_tiler.c > @@ -664,8 +664,9 @@ static int omap_dmm_probe(struct platform_device *dev) > } > > /* set dma mask for device */ > - /* NOTE: this is a workaround for the hwmod not initializing properly */ > - dev->dev.coherent_dma_mask = DMA_BIT_MASK(32); > + ret = dma_set_coherent_mask(&dev->dev, DMA_BIT_MASK(32)); > + if (ret) > + goto fail; Tested with omapdrm on omap4 panda es board. Thanks, Archit

1 0

Re: [alsa-devel] [PATCH v2] sound/soc/fsl/fsl_ssi.c: let check zero instead of check NO_IRQ
by Mark Brown 26 Sep '13

26 Sep '13

On Mon, Sep 23, 2013 at 11:36:21AM +0800, Chen Gang wrote: > NO_IRQ may be defined as '(unsigned int) -1' in some architectures > (arm, sh ...), and either may not be defined in some architectures > which can enable SND_SOC_FSL_SSI (e.g. allmodconfig for arc). Applied, thanks.

2 1

[alsa-devel] [PATCH] HDA-Intel: present up to 8 HDMI/DP outputs via "hdmi" device
by Anssi Hannula 26 Sep '13

26 Sep '13

Some new AMD cards have HDA codecs presenting 6 connected HDMI/DP pin nodes (plus 1 unconnected pin node) according to the ALSA card database. Example: http://www.alsa-project.org/db/?f=de3ced7af41de0ed54d218650e5e2f16c511787b Bump the maximum number of presented HDMI outputs per card via the "hdmi" PCM from 4 to 8 (so that the last possible device is DEV=7). Note that HDMI PCM devices DEV=4..7 use shared PCM device numbers, so HDA cards that have over 4 audio PCM devices or multiple S/PDIF or modem devices will have their remaining PCM devices misrepresented as HDMI devices. Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> --- Another example: http://www.alsa-project.org/db/?f=cd12a9d38f51c47eeedd380e4c5713f0d9a6c9de I do not have confirmed that all of these pins are actually usable (and therefore this patch is untested on affected hardware), but I guess that is probable. Maybe the large amount of them has something to do with DisplayPort supporting multiple displays via a single connector, but that is just a wild guess on my part... But do we want some sort of a solution for the possible conflicts between non-HDMI devices in the 10+ physical PCM device numbering range? src/conf/cards/HDA-Intel.conf | 92 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 92 insertions(+) diff --git a/src/conf/cards/HDA-Intel.conf b/src/conf/cards/HDA-Intel.conf index 3957c12..7976b6c 100644 --- a/src/conf/cards/HDA-Intel.conf +++ b/src/conf/cards/HDA-Intel.conf @@ -305,6 +305,98 @@ HDA-Intel.pcm.hdmi.3 { } } +HDA-Intel.pcm.hdmi.4 { + @args [ CARD AES0 AES1 AES2 AES3 ] + @args.CARD { type string } + @args.AES0 { type integer } + @args.AES1 { type integer } + @args.AES2 { type integer } + @args.AES3 { type integer } + @func refer + name { + @func concat + strings [ + "cards.HDA-Intel.pcm.hdmi.common:" + "CARD=" $CARD "," + "DEVICE=10," + "CTLINDEX=4," + "AES0=" $AES0 "," + "AES1=" $AES1 "," + "AES2=" $AES2 "," + "AES3=" $AES3 + ] + } +} + +HDA-Intel.pcm.hdmi.5 { + @args [ CARD AES0 AES1 AES2 AES3 ] + @args.CARD { type string } + @args.AES0 { type integer } + @args.AES1 { type integer } + @args.AES2 { type integer } + @args.AES3 { type integer } + @func refer + name { + @func concat + strings [ + "cards.HDA-Intel.pcm.hdmi.common:" + "CARD=" $CARD "," + "DEVICE=11," + "CTLINDEX=5," + "AES0=" $AES0 "," + "AES1=" $AES1 "," + "AES2=" $AES2 "," + "AES3=" $AES3 + ] + } +} + +HDA-Intel.pcm.hdmi.6 { + @args [ CARD AES0 AES1 AES2 AES3 ] + @args.CARD { type string } + @args.AES0 { type integer } + @args.AES1 { type integer } + @args.AES2 { type integer } + @args.AES3 { type integer } + @func refer + name { + @func concat + strings [ + "cards.HDA-Intel.pcm.hdmi.common:" + "CARD=" $CARD "," + "DEVICE=12," + "CTLINDEX=6," + "AES0=" $AES0 "," + "AES1=" $AES1 "," + "AES2=" $AES2 "," + "AES3=" $AES3 + ] + } +} + +HDA-Intel.pcm.hdmi.7 { + @args [ CARD AES0 AES1 AES2 AES3 ] + @args.CARD { type string } + @args.AES0 { type integer } + @args.AES1 { type integer } + @args.AES2 { type integer } + @args.AES3 { type integer } + @func refer + name { + @func concat + strings [ + "cards.HDA-Intel.pcm.hdmi.common:" + "CARD=" $CARD "," + "DEVICE=13," + "CTLINDEX=7," + "AES0=" $AES0 "," + "AES1=" $AES1 "," + "AES2=" $AES2 "," + "AES3=" $AES3 + ] + } +} + <confdir:pcm/modem.conf> HDA-Intel.pcm.modem.0 { -- 1.8.1.5

2 2

[alsa-devel] [PATCH 1/1] ALSA: firewire-lib: use inlune function to calculate frame bytes
by Takashi Sakamoto 26 Sep '13

26 Sep '13

Calculating frame bytes can be replaced with inline function in include/sound/pcm.h. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c index ea995af..4b08b25 100644 --- a/sound/firewire/amdtp.c +++ b/sound/firewire/amdtp.c @@ -268,7 +268,7 @@ static void amdtp_write_s32(struct amdtp_out_stream *s, channels = s->pcm_channels; src = (void *)runtime->dma_area + - s->pcm_buffer_pointer * (runtime->frame_bits / 8); + frames_to_bytes(runtime, s->pcm_buffer_pointer); remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer; frame_step = s->data_block_quadlets - channels; @@ -294,7 +294,7 @@ static void amdtp_write_s16(struct amdtp_out_stream *s, channels = s->pcm_channels; src = (void *)runtime->dma_area + - s->pcm_buffer_pointer * (runtime->frame_bits / 8); + frames_to_bytes(runtime, s->pcm_buffer_pointer); remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer; frame_step = s->data_block_quadlets - channels; -- 1.8.1.2

2 1

Re: [alsa-devel] [PATCH 28/51] DMA-API: sound: fix dma mask handling in a lot of drivers
by Takashi Iwai 26 Sep '13

26 Sep '13

Hi, sorry for the lat response, as I've been traveling in the last weeks. At Thu, 19 Sep 2013 22:53:02 +0100, Russell King wrote: > > This code sequence is unsafe in modules: > > static u64 mask = DMA_BIT_MASK(something); > ... > if (!dev->dma_mask) > dev->dma_mask = &mask; > > as if a module is reloaded, the mask will be pointing at the original > module's mask address, and this can lead to oopses. Moreover, they > all follow this with: > > if (!dev->coherent_dma_mask) > dev->coherent_dma_mask = mask; > > where 'mask' is the same value as the statically defined mask, and this > bypasses the architecture's check on whether the DMA mask is possible. > > Fix these issues by using the new dma_coerce_coherent_and_mask() > function. > > Signed-off-by: Russell King <rmk+kernel(a)arm.linux.org.uk> Applied with Mark's ack now. BTW, sound/soc/fsl/imx-pcm-fiq.c wasn't covered by this patch, so I fixed it, too. Thanks! Takashi > --- > sound/arm/pxa2xx-pcm.c | 9 +++------ > sound/soc/atmel/atmel-pcm.c | 11 ++++------- > sound/soc/blackfin/bf5xx-ac97-pcm.c | 11 ++++------- > sound/soc/blackfin/bf5xx-i2s-pcm.c | 10 ++++------ > sound/soc/davinci/davinci-pcm.c | 9 +++------ > sound/soc/fsl/fsl_dma.c | 9 +++------ > sound/soc/fsl/mpc5200_dma.c | 10 ++++------ > sound/soc/jz4740/jz4740-pcm.c | 12 ++++-------- > sound/soc/kirkwood/kirkwood-dma.c | 9 +++------ > sound/soc/nuc900/nuc900-pcm.c | 9 ++++----- > sound/soc/omap/omap-pcm.c | 11 ++++------- > sound/soc/pxa/pxa2xx-pcm.c | 11 ++++------- > sound/soc/s6000/s6000-pcm.c | 9 +++------ > sound/soc/samsung/dma.c | 11 ++++------- > sound/soc/samsung/idma.c | 11 ++++------- > 15 files changed, 55 insertions(+), 97 deletions(-) > > diff --git a/sound/arm/pxa2xx-pcm.c b/sound/arm/pxa2xx-pcm.c > index 69a2455..fb3b76f 100644 > --- a/sound/arm/pxa2xx-pcm.c > +++ b/sound/arm/pxa2xx-pcm.c > @@ -83,8 +83,6 @@ static struct snd_pcm_ops pxa2xx_pcm_ops = { > .mmap = pxa2xx_pcm_mmap, > }; > > -static u64 pxa2xx_pcm_dmamask = 0xffffffff; > - > int pxa2xx_pcm_new(struct snd_card *card, struct pxa2xx_pcm_client *client, > struct snd_pcm **rpcm) > { > @@ -100,10 +98,9 @@ int pxa2xx_pcm_new(struct snd_card *card, struct pxa2xx_pcm_client *client, > pcm->private_data = client; > pcm->private_free = pxa2xx_pcm_free_dma_buffers; > > - if (!card->dev->dma_mask) > - card->dev->dma_mask = &pxa2xx_pcm_dmamask; > - if (!card->dev->coherent_dma_mask) > - card->dev->coherent_dma_mask = 0xffffffff; > + ret = dma_coerce_mask_and_coherent_mask(card->dev, DMA_BIT_MASK(32)); > + if (ret) > + goto out; > > if (play) { > int stream = SNDRV_PCM_STREAM_PLAYBACK; > diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c > index 3109db7..fbb87e3 100644 > --- a/sound/soc/atmel/atmel-pcm.c > +++ b/sound/soc/atmel/atmel-pcm.c > @@ -68,18 +68,15 @@ int atmel_pcm_mmap(struct snd_pcm_substream *substream, > } > EXPORT_SYMBOL_GPL(atmel_pcm_mmap); > > -static u64 atmel_pcm_dmamask = DMA_BIT_MASK(32); > - > int atmel_pcm_new(struct snd_soc_pcm_runtime *rtd) > { > struct snd_card *card = rtd->card->snd_card; > struct snd_pcm *pcm = rtd->pcm; > - int ret = 0; > + int ret; > > - if (!card->dev->dma_mask) > - card->dev->dma_mask = &atmel_pcm_dmamask; > - if (!card->dev->coherent_dma_mask) > - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); > + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); > + if (ret) > + return ret; > > if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { > pr_debug("atmel-pcm: allocating PCM playback DMA buffer\n"); > diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c > index 53f8408..1d4c676 100644 > --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c > +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c > @@ -415,19 +415,16 @@ static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm) > } > } > > -static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32); > - > static int bf5xx_pcm_ac97_new(struct snd_soc_pcm_runtime *rtd) > { > struct snd_card *card = rtd->card->snd_card; > struct snd_pcm *pcm = rtd->pcm; > - int ret = 0; > + int ret; > > pr_debug("%s enter\n", __func__); > - if (!card->dev->dma_mask) > - card->dev->dma_mask = &bf5xx_pcm_dmamask; > - if (!card->dev->coherent_dma_mask) > - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); > + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); > + if (ret) > + return ret; > > if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { > ret = bf5xx_pcm_preallocate_dma_buffer(pcm, > diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c > index 9cb4a80..2a5b434 100644 > --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c > +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c > @@ -323,18 +323,16 @@ static struct snd_pcm_ops bf5xx_pcm_i2s_ops = { > .silence = bf5xx_pcm_silence, > }; > > -static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32); > - > static int bf5xx_pcm_i2s_new(struct snd_soc_pcm_runtime *rtd) > { > struct snd_card *card = rtd->card->snd_card; > size_t size = bf5xx_pcm_hardware.buffer_bytes_max; > + int ret; > > pr_debug("%s enter\n", __func__); > - if (!card->dev->dma_mask) > - card->dev->dma_mask = &bf5xx_pcm_dmamask; > - if (!card->dev->coherent_dma_mask) > - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); > + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); > + if (ret) > + return ret; > > return snd_pcm_lib_preallocate_pages_for_all(rtd->pcm, > SNDRV_DMA_TYPE_DEV, card->dev, size, size); > diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c > index 8460edc..84a63c6 100644 > --- a/sound/soc/davinci/davinci-pcm.c > +++ b/sound/soc/davinci/davinci-pcm.c > @@ -844,18 +844,15 @@ static void davinci_pcm_free(struct snd_pcm *pcm) > } > } > > -static u64 davinci_pcm_dmamask = DMA_BIT_MASK(32); > - > static int davinci_pcm_new(struct snd_soc_pcm_runtime *rtd) > { > struct snd_card *card = rtd->card->snd_card; > struct snd_pcm *pcm = rtd->pcm; > int ret; > > - if (!card->dev->dma_mask) > - card->dev->dma_mask = &davinci_pcm_dmamask; > - if (!card->dev->coherent_dma_mask) > - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); > + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); > + if (ret) > + return ret; > > if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { > ret = davinci_pcm_preallocate_dma_buffer(pcm, > diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c > index 9cc5c1f..f73c7ef 100644 > --- a/sound/soc/fsl/fsl_dma.c > +++ b/sound/soc/fsl/fsl_dma.c > @@ -298,14 +298,11 @@ static int fsl_dma_new(struct snd_soc_pcm_runtime *rtd) > { > struct snd_card *card = rtd->card->snd_card; > struct snd_pcm *pcm = rtd->pcm; > - static u64 fsl_dma_dmamask = DMA_BIT_MASK(36); > int ret; > > - if (!card->dev->dma_mask) > - card->dev->dma_mask = &fsl_dma_dmamask; > - > - if (!card->dev->coherent_dma_mask) > - card->dev->coherent_dma_mask = fsl_dma_dmamask; > + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(36)); > + if (ret) > + return ret; > > /* Some codecs have separate DAIs for playback and capture, so we > * should allocate a DMA buffer only for the streams that are valid. > diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c > index 2a847ca..8fcf224 100644 > --- a/sound/soc/fsl/mpc5200_dma.c > +++ b/sound/soc/fsl/mpc5200_dma.c > @@ -299,7 +299,6 @@ static struct snd_pcm_ops psc_dma_ops = { > .hw_params = psc_dma_hw_params, > }; > > -static u64 psc_dma_dmamask = DMA_BIT_MASK(32); > static int psc_dma_new(struct snd_soc_pcm_runtime *rtd) > { > struct snd_card *card = rtd->card->snd_card; > @@ -307,15 +306,14 @@ static int psc_dma_new(struct snd_soc_pcm_runtime *rtd) > struct snd_pcm *pcm = rtd->pcm; > struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai); > size_t size = psc_dma_hardware.buffer_bytes_max; > - int rc = 0; > + int rc; > > dev_dbg(rtd->platform->dev, "psc_dma_new(card=%p, dai=%p, pcm=%p)\n", > card, dai, pcm); > > - if (!card->dev->dma_mask) > - card->dev->dma_mask = &psc_dma_dmamask; > - if (!card->dev->coherent_dma_mask) > - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); > + rc = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); > + if (rc) > + return rc; > > if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { > rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->card->dev, > diff --git a/sound/soc/jz4740/jz4740-pcm.c b/sound/soc/jz4740/jz4740-pcm.c > index 7100592..1d7ef28 100644 > --- a/sound/soc/jz4740/jz4740-pcm.c > +++ b/sound/soc/jz4740/jz4740-pcm.c > @@ -297,19 +297,15 @@ static void jz4740_pcm_free(struct snd_pcm *pcm) > } > } > > -static u64 jz4740_pcm_dmamask = DMA_BIT_MASK(32); > - > static int jz4740_pcm_new(struct snd_soc_pcm_runtime *rtd) > { > struct snd_card *card = rtd->card->snd_card; > struct snd_pcm *pcm = rtd->pcm; > - int ret = 0; > - > - if (!card->dev->dma_mask) > - card->dev->dma_mask = &jz4740_pcm_dmamask; > + int ret; > > - if (!card->dev->coherent_dma_mask) > - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); > + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); > + if (ret) > + return ret; > > if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { > ret = jz4740_pcm_preallocate_dma_buffer(pcm, > diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c > index b238434..3814bb0 100644 > --- a/sound/soc/kirkwood/kirkwood-dma.c > +++ b/sound/soc/kirkwood/kirkwood-dma.c > @@ -59,8 +59,6 @@ static struct snd_pcm_hardware kirkwood_dma_snd_hw = { > .fifo_size = 0, > }; > > -static u64 kirkwood_dma_dmamask = DMA_BIT_MASK(32); > - > static irqreturn_t kirkwood_dma_irq(int irq, void *dev_id) > { > struct kirkwood_dma_data *priv = dev_id; > @@ -292,10 +290,9 @@ static int kirkwood_dma_new(struct snd_soc_pcm_runtime *rtd) > struct snd_pcm *pcm = rtd->pcm; > int ret; > > - if (!card->dev->dma_mask) > - card->dev->dma_mask = &kirkwood_dma_dmamask; > - if (!card->dev->coherent_dma_mask) > - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); > + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); > + if (ret) > + return ret; > > if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { > ret = kirkwood_dma_preallocate_dma_buffer(pcm, > diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c > index c894ff0..f588ee4 100644 > --- a/sound/soc/nuc900/nuc900-pcm.c > +++ b/sound/soc/nuc900/nuc900-pcm.c > @@ -314,16 +314,15 @@ static void nuc900_dma_free_dma_buffers(struct snd_pcm *pcm) > snd_pcm_lib_preallocate_free_for_all(pcm); > } > > -static u64 nuc900_pcm_dmamask = DMA_BIT_MASK(32); > static int nuc900_dma_new(struct snd_soc_pcm_runtime *rtd) > { > struct snd_card *card = rtd->card->snd_card; > struct snd_pcm *pcm = rtd->pcm; > + int ret; > > - if (!card->dev->dma_mask) > - card->dev->dma_mask = &nuc900_pcm_dmamask; > - if (!card->dev->coherent_dma_mask) > - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); > + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); > + if (ret) > + return ret; > > snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, > card->dev, 4 * 1024, (4 * 1024) - 1); > diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c > index a11405d..b8fa986 100644 > --- a/sound/soc/omap/omap-pcm.c > +++ b/sound/soc/omap/omap-pcm.c > @@ -156,8 +156,6 @@ static struct snd_pcm_ops omap_pcm_ops = { > .mmap = omap_pcm_mmap, > }; > > -static u64 omap_pcm_dmamask = DMA_BIT_MASK(64); > - > static int omap_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, > int stream) > { > @@ -202,12 +200,11 @@ static int omap_pcm_new(struct snd_soc_pcm_runtime *rtd) > { > struct snd_card *card = rtd->card->snd_card; > struct snd_pcm *pcm = rtd->pcm; > - int ret = 0; > + int ret; > > - if (!card->dev->dma_mask) > - card->dev->dma_mask = &omap_pcm_dmamask; > - if (!card->dev->coherent_dma_mask) > - card->dev->coherent_dma_mask = DMA_BIT_MASK(64); > + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(64)); > + if (ret) > + return ret; > > if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { > ret = omap_pcm_preallocate_dma_buffer(pcm, > diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c > index 806da27..d58b09f 100644 > --- a/sound/soc/pxa/pxa2xx-pcm.c > +++ b/sound/soc/pxa/pxa2xx-pcm.c > @@ -87,18 +87,15 @@ static struct snd_pcm_ops pxa2xx_pcm_ops = { > .mmap = pxa2xx_pcm_mmap, > }; > > -static u64 pxa2xx_pcm_dmamask = DMA_BIT_MASK(32); > - > static int pxa2xx_soc_pcm_new(struct snd_soc_pcm_runtime *rtd) > { > struct snd_card *card = rtd->card->snd_card; > struct snd_pcm *pcm = rtd->pcm; > - int ret = 0; > + int ret; > > - if (!card->dev->dma_mask) > - card->dev->dma_mask = &pxa2xx_pcm_dmamask; > - if (!card->dev->coherent_dma_mask) > - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); > + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); > + if (ret) > + return ret; > > if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { > ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, > diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c > index d0740a7..283620a 100644 > --- a/sound/soc/s6000/s6000-pcm.c > +++ b/sound/soc/s6000/s6000-pcm.c > @@ -444,8 +444,6 @@ static void s6000_pcm_free(struct snd_pcm *pcm) > snd_pcm_lib_preallocate_free_for_all(pcm); > } > > -static u64 s6000_pcm_dmamask = DMA_BIT_MASK(32); > - > static int s6000_pcm_new(struct snd_soc_pcm_runtime *runtime) > { > struct snd_card *card = runtime->card->snd_card; > @@ -456,10 +454,9 @@ static int s6000_pcm_new(struct snd_soc_pcm_runtime *runtime) > params = snd_soc_dai_get_dma_data(runtime->cpu_dai, > pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream); > > - if (!card->dev->dma_mask) > - card->dev->dma_mask = &s6000_pcm_dmamask; > - if (!card->dev->coherent_dma_mask) > - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); > + res = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); > + if (res) > + return res; > > if (params->dma_in) { > s6dmac_disable_chan(DMA_MASK_DMAC(params->dma_in), > diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c > index 9338d11..fe2748b 100644 > --- a/sound/soc/samsung/dma.c > +++ b/sound/soc/samsung/dma.c > @@ -406,20 +406,17 @@ static void dma_free_dma_buffers(struct snd_pcm *pcm) > } > } > > -static u64 dma_mask = DMA_BIT_MASK(32); > - > static int dma_new(struct snd_soc_pcm_runtime *rtd) > { > struct snd_card *card = rtd->card->snd_card; > struct snd_pcm *pcm = rtd->pcm; > - int ret = 0; > + int ret; > > pr_debug("Entered %s\n", __func__); > > - if (!card->dev->dma_mask) > - card->dev->dma_mask = &dma_mask; > - if (!card->dev->coherent_dma_mask) > - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); > + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); > + if (ret) > + return ret; > > if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { > ret = preallocate_dma_buffer(pcm, > diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c > index ce1e1e1..e4f318f 100644 > --- a/sound/soc/samsung/idma.c > +++ b/sound/soc/samsung/idma.c > @@ -383,18 +383,15 @@ static int preallocate_idma_buffer(struct snd_pcm *pcm, int stream) > return 0; > } > > -static u64 idma_mask = DMA_BIT_MASK(32); > - > static int idma_new(struct snd_soc_pcm_runtime *rtd) > { > struct snd_card *card = rtd->card->snd_card; > struct snd_pcm *pcm = rtd->pcm; > - int ret = 0; > + int ret; > > - if (!card->dev->dma_mask) > - card->dev->dma_mask = &idma_mask; > - if (!card->dev->coherent_dma_mask) > - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); > + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); > + if (ret) > + return ret; > > if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { > ret = preallocate_idma_buffer(pcm, > -- > 1.7.4.4 >

2 3

Re: [alsa-devel] Cannot load snd-usb-audio on 3.12-rc2
by Daniel Mack 26 Sep '13

26 Sep '13

Hi Sarah, On 24.09.2013 01:41, Sarah Sharp wrote: > The snd-usb-audio driver no longer loads properly on 3.12-rc2 when I > plug in my USB headset. It worked fine on 3.11-rc4 (not sure about > vanilla 3.11). > > Trying to manually load the driver fails: > > sarah@xanatos:~$ sudo modprobe snd-usb-audio > FATAL: Error inserting snd_usb_audio > (/lib/modules/3.12.0-rc2/kernel/sound/usb/snd-usb-audio.ko): Invalid argument > > dmesg shows: > > [ 44.231668] snd_usb_audio: `-2' invalid for parameter `index' That's really strange. I just tested on an ARM board where I currently run 3.12-rc2 as well, and I can load the module without problems. Also, sound/usb/card.c (where the params are declared) hasn't seen an update in 3.12. I suspect your OS is passing "index=-2" from something like /etc/modprobe.d/* or cmdline settings. Could you investigate in that direction? HTH, Daniel

2 1

[alsa-devel] [PATCH] ALSA: improve buffer size computations for USB PCM audio
by Alan Stern 26 Sep '13

26 Sep '13

This patch changes the way URBs are allocated and their sizes are determined for PCM playback in the snd-usb-audio driver. Currently the driver allocates too few URBs for endpoints that don't use implicit sync, making underruns more likely to occur. This may be a holdover from before I/O delays could be measured accurately; in any case, it is no longer necessary. The patch allocates as many URBs as possible, subject to four limitations: The total number of URBs for the endpoint is not allowed to exceed MAX_URBS (which the patch increases from 8 to 12). The total number of packets per URB is not allowed to exceed MAX_PACKS (or MAX_PACKS_HS for high-speed devices), which is decreased from 20 to 6. The total duration of queued data is not allowed to exceed MAX_QUEUE, which is decreased from 24 ms to 18 ms. The total number of ALSA frames in the output queue is not allowed to exceed the ALSA buffer size. The last requirement is the hardest to implement. Currently the number of URBs needed to fill a buffer cannot be determined in advance, because a buffer contains a fixed number of frames whereas the number of frames in an URB varies to match shifts in the device's clock rate. To solve this problem, the patch changes the logic for deciding how many packets an URB should contain. Rather than using as many as possible without exceeding an ALSA period boundary, now the driver uses only as many packets as needed to transfer a predetermined number of frames. As a result, unless the device's clock has an exceedingly variable rate, the number of URBs making up each period (and hence each buffer) will remain constant. The overall effect of the patch is that playback works better in low-latency settings. The user can still specify values for frames/period and periods/buffer that exceed the capabilities of the hardware, of course. But for values that are within those capabilities, the performance will be improved. For example, testing shows that a high-speed device can handle 32 frames/period and 3 periods/buffer at 48 KHz, whereas the current driver starts to get glitchy at 64 frames/period and 2 periods/buffer. A side effect of these changes is that the "nrpacks" module parameter is no longer used. The patch removes it. Signed-off-by: Alan Stern <stern(a)rowland.harvard.edu> CC: Clemens Ladisch <clemens(a)ladisch.de> Tested-by: Daniel Mack <zonque(a)gmail.com> Tested-by: Eldad Zack <eldad(a)fogrefinery.com> --- [as1712] sound/usb/card.c | 8 --- sound/usb/card.h | 10 +++- sound/usb/endpoint.c | 108 +++++++++++++++++++++++++++++---------------------- sound/usb/endpoint.h | 2 sound/usb/pcm.c | 14 +++++- sound/usb/usbaudio.h | 1 6 files changed, 83 insertions(+), 60 deletions(-) Index: usb-3.11/sound/usb/usbaudio.h =================================================================== --- usb-3.11.orig/sound/usb/usbaudio.h +++ usb-3.11/sound/usb/usbaudio.h @@ -55,7 +55,6 @@ struct snd_usb_audio { struct list_head mixer_list; /* list of mixer interfaces */ int setup; /* from the 'device_setup' module param */ - int nrpacks; /* from the 'nrpacks' module param */ bool autoclock; /* from the 'autoclock' module param */ struct usb_host_interface *ctrl_intf; /* the audio control interface */ Index: usb-3.11/sound/usb/card.c =================================================================== --- usb-3.11.orig/sound/usb/card.c +++ usb-3.11/sound/usb/card.c @@ -79,7 +79,6 @@ static bool enable[SNDRV_CARDS] = SNDRV_ /* Vendor/product IDs for this card */ static int vid[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = -1 }; static int pid[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = -1 }; -static int nrpacks = 8; /* max. number of packets per urb */ static int device_setup[SNDRV_CARDS]; /* device parameter for this card */ static bool ignore_ctl_error; static bool autoclock = true; @@ -94,8 +93,6 @@ module_param_array(vid, int, NULL, 0444) MODULE_PARM_DESC(vid, "Vendor ID for the USB audio device."); module_param_array(pid, int, NULL, 0444); MODULE_PARM_DESC(pid, "Product ID for the USB audio device."); -module_param(nrpacks, int, 0644); -MODULE_PARM_DESC(nrpacks, "Max. number of packets per URB."); module_param_array(device_setup, int, NULL, 0444); MODULE_PARM_DESC(device_setup, "Specific device setup (if needed)."); module_param(ignore_ctl_error, bool, 0444); @@ -374,7 +371,6 @@ static int snd_usb_audio_create(struct u chip->dev = dev; chip->card = card; chip->setup = device_setup[idx]; - chip->nrpacks = nrpacks; chip->autoclock = autoclock; chip->probing = 1; @@ -756,10 +752,6 @@ static struct usb_driver usb_audio_drive static int __init snd_usb_audio_init(void) { - if (nrpacks < 1 || nrpacks > MAX_PACKS) { - printk(KERN_WARNING "invalid nrpacks value.\n"); - return -EINVAL; - } return usb_register(&usb_audio_driver); } Index: usb-3.11/sound/usb/endpoint.h =================================================================== --- usb-3.11.orig/sound/usb/endpoint.h +++ usb-3.11/sound/usb/endpoint.h @@ -12,6 +12,8 @@ int snd_usb_endpoint_set_params(struct s snd_pcm_format_t pcm_format, unsigned int channels, unsigned int period_bytes, + unsigned int period_frames, + unsigned int buffer_periods, unsigned int rate, struct audioformat *fmt, struct snd_usb_endpoint *sync_ep); Index: usb-3.11/sound/usb/card.h =================================================================== --- usb-3.11.orig/sound/usb/card.h +++ usb-3.11/sound/usb/card.h @@ -2,11 +2,11 @@ #define __USBAUDIO_CARD_H #define MAX_NR_RATES 1024 -#define MAX_PACKS 20 +#define MAX_PACKS 6 /* per URB */ #define MAX_PACKS_HS (MAX_PACKS * 8) /* in high speed mode */ -#define MAX_URBS 8 +#define MAX_URBS 12 #define SYNC_URBS 4 /* always four urbs for sync */ -#define MAX_QUEUE 24 /* try not to exceed this queue length, in ms */ +#define MAX_QUEUE 18 /* try not to exceed this queue length, in ms */ struct audioformat { struct list_head list; @@ -87,6 +87,7 @@ struct snd_usb_endpoint { unsigned int phase; /* phase accumulator */ unsigned int maxpacksize; /* max packet size in bytes */ unsigned int maxframesize; /* max packet size in frames */ + unsigned int max_urb_frames; /* max URB size in frames */ unsigned int curpacksize; /* current packet size in bytes (for capture) */ unsigned int curframesize; /* current packet size in frames (for capture) */ unsigned int syncmaxsize; /* sync endpoint packet size */ @@ -116,6 +117,8 @@ struct snd_usb_substream { unsigned int channels_max; /* max channels in the all audiofmts */ unsigned int cur_rate; /* current rate (for hw_params callback) */ unsigned int period_bytes; /* current period bytes (for hw_params callback) */ + unsigned int period_frames; /* current frames per period */ + unsigned int buffer_periods; /* current periods per buffer */ unsigned int altset_idx; /* USB data format: index of alternate setting */ unsigned int txfr_quirk:1; /* allow sub-frame alignment */ unsigned int fmt_type; /* USB audio format type (1-3) */ @@ -125,6 +128,7 @@ struct snd_usb_substream { unsigned int hwptr_done; /* processed byte position in the buffer */ unsigned int transfer_done; /* processed frames since last period update */ + unsigned int frame_limit; /* limits number of packets in URB */ /* data and sync endpoints for this stream */ unsigned int ep_num; /* the endpoint number */ Index: usb-3.11/sound/usb/pcm.c =================================================================== --- usb-3.11.orig/sound/usb/pcm.c +++ usb-3.11/sound/usb/pcm.c @@ -562,6 +562,7 @@ static int configure_sync_endpoint(struc subs->pcm_format, subs->channels, subs->period_bytes, + 0, 0, subs->cur_rate, subs->cur_audiofmt, NULL); @@ -598,6 +599,7 @@ static int configure_sync_endpoint(struc subs->pcm_format, sync_fp->channels, sync_period_bytes, + 0, 0, subs->cur_rate, sync_fp, NULL); @@ -620,6 +622,8 @@ static int configure_endpoint(struct snd subs->pcm_format, subs->channels, subs->period_bytes, + subs->period_frames, + subs->buffer_periods, subs->cur_rate, subs->cur_audiofmt, subs->sync_endpoint); @@ -656,6 +660,8 @@ static int snd_usb_hw_params(struct snd_ subs->pcm_format = params_format(hw_params); subs->period_bytes = params_period_bytes(hw_params); + subs->period_frames = params_period_size(hw_params); + subs->buffer_periods = params_periods(hw_params); subs->channels = params_channels(hw_params); subs->cur_rate = params_rate(hw_params); @@ -1330,6 +1336,7 @@ static void prepare_playback_urb(struct frames = 0; urb->number_of_packets = 0; spin_lock_irqsave(&subs->lock, flags); + subs->frame_limit += ep->max_urb_frames; for (i = 0; i < ctx->packets; i++) { if (ctx->packet_size[i]) counts = ctx->packet_size[i]; @@ -1344,6 +1351,7 @@ static void prepare_playback_urb(struct subs->transfer_done += counts; if (subs->transfer_done >= runtime->period_size) { subs->transfer_done -= runtime->period_size; + subs->frame_limit = 0; period_elapsed = 1; if (subs->fmt_type == UAC_FORMAT_TYPE_II) { if (subs->transfer_done > 0) { @@ -1366,8 +1374,10 @@ static void prepare_playback_urb(struct break; } } - if (period_elapsed && - !snd_usb_endpoint_implicit_feedback_sink(subs->data_endpoint)) /* finish at the period boundary */ + /* finish at the period boundary or after enough frames */ + if ((period_elapsed || + subs->transfer_done >= subs->frame_limit) && + !snd_usb_endpoint_implicit_feedback_sink(ep)) break; } bytes = frames * ep->stride; Index: usb-3.11/sound/usb/endpoint.c =================================================================== --- usb-3.11.orig/sound/usb/endpoint.c +++ usb-3.11/sound/usb/endpoint.c @@ -571,11 +571,14 @@ static int data_ep_set_params(struct snd snd_pcm_format_t pcm_format, unsigned int channels, unsigned int period_bytes, + unsigned int frames_per_period, + unsigned int periods_per_buffer, struct audioformat *fmt, struct snd_usb_endpoint *sync_ep) { - unsigned int maxsize, i, urb_packs, total_packs, packs_per_ms; - int is_playback = usb_pipeout(ep->pipe); + unsigned int maxsize, minsize, packs_per_ms, max_packs_per_urb; + unsigned int max_packs_per_period, urbs_per_period, urb_packs; + unsigned int max_urbs, i; int frame_bits = snd_pcm_format_physical_width(pcm_format) * channels; if (pcm_format == SNDRV_PCM_FORMAT_DSD_U16_LE && fmt->dsd_dop) { @@ -608,58 +611,67 @@ static int data_ep_set_params(struct snd else ep->curpacksize = maxsize; - if (snd_usb_get_speed(ep->chip->dev) != USB_SPEED_FULL) + if (snd_usb_get_speed(ep->chip->dev) != USB_SPEED_FULL) { packs_per_ms = 8 >> ep->datainterval; - else - packs_per_ms = 1; - - if (is_playback && !snd_usb_endpoint_implicit_feedback_sink(ep)) { - urb_packs = max(ep->chip->nrpacks, 1); - urb_packs = min(urb_packs, (unsigned int) MAX_PACKS); + max_packs_per_urb = MAX_PACKS_HS; } else { - urb_packs = 1; + packs_per_ms = 1; + max_packs_per_urb = MAX_PACKS; } - - urb_packs *= packs_per_ms; - if (sync_ep && !snd_usb_endpoint_implicit_feedback_sink(ep)) - urb_packs = min(urb_packs, 1U << sync_ep->syncinterval); + max_packs_per_urb = min(max_packs_per_urb, + 1U << sync_ep->syncinterval); + max_packs_per_urb = max(1u, max_packs_per_urb >> ep->datainterval); + + /* + * Capture endpoints need to use small URBs because there's no way + * to tell in advance where the next period will end, and we don't + * want the next URB to complete much after the period ends. + * + * Playback endpoints with implicit sync much use the same parameters + * as their corresponding capture endpoint. + */ + if (usb_pipein(ep->pipe) || + snd_usb_endpoint_implicit_feedback_sink(ep)) { + + /* make capture URBs <= 1 ms and smaller than a period */ + urb_packs = min(max_packs_per_urb, packs_per_ms); + while (urb_packs > 1 && urb_packs * maxsize >= period_bytes) + urb_packs >>= 1; + ep->nurbs = MAX_URBS; - /* decide how many packets to be used */ - if (is_playback && !snd_usb_endpoint_implicit_feedback_sink(ep)) { - unsigned int minsize, maxpacks; + /* + * Playback endpoints without implicit sync are adjusted so that + * a period fits as evenly as possible in the smallest number of + * URBs. The total number of URBs is adjusted to the size of the + * ALSA buffer, subject to the MAX_URBS and MAX_QUEUE limits. + */ + } else { /* determine how small a packet can be */ - minsize = (ep->freqn >> (16 - ep->datainterval)) - * (frame_bits >> 3); + minsize = (ep->freqn >> (16 - ep->datainterval)) * + (frame_bits >> 3); /* with sync from device, assume it can be 12% lower */ if (sync_ep) minsize -= minsize >> 3; minsize = max(minsize, 1u); - total_packs = (period_bytes + minsize - 1) / minsize; - /* we need at least two URBs for queueing */ - if (total_packs < 2) { - total_packs = 2; - } else { - /* and we don't want too long a queue either */ - maxpacks = max(MAX_QUEUE * packs_per_ms, urb_packs * 2); - total_packs = min(total_packs, maxpacks); - } - } else { - while (urb_packs > 1 && urb_packs * maxsize >= period_bytes) - urb_packs >>= 1; - total_packs = MAX_URBS * urb_packs; - } - ep->nurbs = (total_packs + urb_packs - 1) / urb_packs; - if (ep->nurbs > MAX_URBS) { - /* too much... */ - ep->nurbs = MAX_URBS; - total_packs = MAX_URBS * urb_packs; - } else if (ep->nurbs < 2) { - /* too little - we need at least two packets - * to ensure contiguous playback/capture - */ - ep->nurbs = 2; + /* how many packets will contain an entire ALSA period? */ + max_packs_per_period = DIV_ROUND_UP(period_bytes, minsize); + + /* how many URBs will contain a period? */ + urbs_per_period = DIV_ROUND_UP(max_packs_per_period, + max_packs_per_urb); + /* how many packets are needed in each URB? */ + urb_packs = DIV_ROUND_UP(max_packs_per_period, urbs_per_period); + + /* limit the number of frames in a single URB */ + ep->max_urb_frames = DIV_ROUND_UP(frames_per_period, + urbs_per_period); + + /* try to use enough URBs to contain an entire ALSA buffer */ + max_urbs = min((unsigned) MAX_URBS, + MAX_QUEUE * packs_per_ms / urb_packs); + ep->nurbs = min(max_urbs, urbs_per_period * periods_per_buffer); } /* allocate and initialize data urbs */ @@ -667,8 +679,7 @@ static int data_ep_set_params(struct snd struct snd_urb_ctx *u = &ep->urb[i]; u->index = i; u->ep = ep; - u->packets = (i + 1) * total_packs / ep->nurbs - - i * total_packs / ep->nurbs; + u->packets = urb_packs; u->buffer_size = maxsize * u->packets; if (fmt->fmt_type == UAC_FORMAT_TYPE_II) @@ -745,6 +756,8 @@ out_of_memory: * @pcm_format: the audio fomat. * @channels: the number of audio channels. * @period_bytes: the number of bytes in one alsa period. + * @period_frames: the number of frames in one alsa period. + * @buffer_periods: the number of periods in one alsa buffer. * @rate: the frame rate. * @fmt: the USB audio format information * @sync_ep: the sync endpoint to use, if any @@ -757,6 +770,8 @@ int snd_usb_endpoint_set_params(struct s snd_pcm_format_t pcm_format, unsigned int channels, unsigned int period_bytes, + unsigned int period_frames, + unsigned int buffer_periods, unsigned int rate, struct audioformat *fmt, struct snd_usb_endpoint *sync_ep) @@ -790,7 +805,8 @@ int snd_usb_endpoint_set_params(struct s switch (ep->type) { case SND_USB_ENDPOINT_TYPE_DATA: err = data_ep_set_params(ep, pcm_format, channels, - period_bytes, fmt, sync_ep); + period_bytes, period_frames, + buffer_periods, fmt, sync_ep); break; case SND_USB_ENDPOINT_TYPE_SYNC: err = sync_ep_set_params(ep, fmt);

2 1

[alsa-devel] How to deal with snooping in firmware?
by Paul Menzel 26 Sep '13

26 Sep '13

Dear ALSA folks, with Ubuntu 13.04 and AMD Kabini boards, there are audio distortions, which are not present under Microsoft Windows [1][2]. Disabling snooping fixes this, which is in line with the information from an earlier discussion [3]. As it works under Microsoft Windows, is this indeed a driver problem and should be fixed there? Is it better to disabling snooping by default in the firmware and let the driver enable it? What do you propose how to fix this? Thanks, Paul [1] http://review.coreboot.org/#/c/3934/ [2] http://review.coreboot.org/#/c/3934/1/src/mainboard/amd/olivehill/get_bus_c… [3] http://lists.freedesktop.org/archives/pulseaudio-discuss/2012-May/013568.ht… [pulseaudio-discuss] [alsa-devel] Adjust volume may cause audio playback corruption

2 1

Re: [alsa-devel] Bug: Conexant CX20588, Kernel 3.11, Win8 changes GPIO settings
by Takashi Iwai 26 Sep '13

26 Sep '13

At Sun, 22 Sep 2013 21:33:59 +0200, Christopher wrote: > > Hi, > > I recently installed Windows 8 parallel to linux and there is no sound in > linux after rebooting. I guess the win8 conexant driver messes something up. > > A quick fix is to change this in HDAAnalyzer: > > GPIO: io=4, o=0, i=0, unsolicited=1, wake=0 > IO[0]: enable=1, dir=1, wake=0, sticky=0, data=0, unsol=0 > IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > > Either set enable=0 or data=1 -> Internal Audio + Headphones work again. > This is persistent between reboots. Booting Win8 resets this again. > > I would be very happy to test a patch. So the GPIO should have been adjusted by the driver from the beginning. Could you give your hardware details? At best, give the output of alsa-info.sh. Run it with --no-upload option and attach the output. thanks, Takashi

1 0

[alsa-devel] [PATCH] ALSA: compress: Make sure we trigger STOP before closing the stream.
by Liam Girdwood 26 Sep '13

26 Sep '13

Currently we assume that userspace will shut down the compressed stream correctly. However, if userspcae dies (e.g. cplay & ctrl-C) we dont stop the stream before freeing it. This now checks that the stream is stopped before freeing. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/core/compress_offload.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 99db892..a9ae4f3 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -139,6 +139,18 @@ static int snd_compr_open(struct inode *inode, struct file *f) static int snd_compr_free(struct inode *inode, struct file *f) { struct snd_compr_file *data = f->private_data; + struct snd_compr_runtime *runtime = data->stream.runtime; + + switch (runtime->state) { + case SNDRV_PCM_STATE_RUNNING: + case SNDRV_PCM_STATE_DRAINING: + case SNDRV_PCM_STATE_PAUSED: + data->stream.ops->trigger(&data->stream, SNDRV_PCM_TRIGGER_STOP); + break; + default: + break; + } + data->stream.ops->free(&data->stream); kfree(data->stream.runtime->buffer); kfree(data->stream.runtime); -- 1.8.1.2

4 10

Re: [alsa-devel] [PATCH 4/5] ALSA: Fix assignment of 0/1 to bool variables
by Takashi Iwai 26 Sep '13

26 Sep '13

At Sun, 22 Sep 2013 20:44:12 +0200, Peter Senna Tschudin wrote: > > Convert 0 to false and 1 to true when assigning values to bool > variables. Inspired by commit 3db1cd5c05f35fb43eb134df6f321de4e63141f2. > > The simplified semantic patch that find this problem is as > follows (http://coccinelle.lip6.fr/): > > @@ > bool b; > @@ > ( > -b = 0 > +b = false > | > -b = 1 > +b = true > ) > > Signed-off-by: Peter Senna Tschudin <peter.senna(a)gmail.com> Thanks, applied now. Takashi > > --- > sound/pci/azt3328.c | 14 +++++++------- > sound/usb/mixer.c | 4 ++-- > 2 files changed, 9 insertions(+), 9 deletions(-) > > diff -u -p a/sound/usb/mixer.c b/sound/usb/mixer.c > --- a/sound/usb/mixer.c > +++ b/sound/usb/mixer.c > @@ -1151,14 +1151,14 @@ static void check_no_speaker_on_headset( > const char *names_to_check[] = { > "Headset", "headset", "Headphone", "headphone", NULL}; > const char **s; > - bool found = 0; > + bool found = false; > > if (strcmp("Speaker", kctl->id.name)) > return; > > for (s = names_to_check; *s; s++) > if (strstr(card->shortname, *s)) { > - found = 1; > + found = true; > break; > } > > diff -u -p a/sound/pci/azt3328.c b/sound/pci/azt3328.c > --- a/sound/pci/azt3328.c > +++ b/sound/pci/azt3328.c > @@ -715,14 +715,14 @@ snd_azf3328_mixer_ac97_read(struct snd_a > const struct snd_azf3328 *chip = ac97->private_data; > unsigned short reg_azf = snd_azf3328_mixer_ac97_map_reg_idx(reg_ac97); > unsigned short reg_val = 0; > - bool unsupported = 0; > + bool unsupported = false; > > snd_azf3328_dbgmixer( > "snd_azf3328_mixer_ac97_read reg_ac97 %u\n", > reg_ac97 > ); > if (reg_azf & AZF_AC97_REG_UNSUPPORTED) > - unsupported = 1; > + unsupported = true; > else { > if (reg_azf & AZF_AC97_REG_REAL_IO_READ) > reg_val = snd_azf3328_mixer_inw(chip, > @@ -759,7 +759,7 @@ snd_azf3328_mixer_ac97_read(struct snd_a > reg_val = azf_emulated_ac97_vendor_id & 0xffff; > break; > default: > - unsupported = 1; > + unsupported = true; > break; > } > } > @@ -776,14 +776,14 @@ snd_azf3328_mixer_ac97_write(struct snd_ > { > const struct snd_azf3328 *chip = ac97->private_data; > unsigned short reg_azf = snd_azf3328_mixer_ac97_map_reg_idx(reg_ac97); > - bool unsupported = 0; > + bool unsupported = false; > > snd_azf3328_dbgmixer( > "snd_azf3328_mixer_ac97_write reg_ac97 %u val %u\n", > reg_ac97, val > ); > if (reg_azf & AZF_AC97_REG_UNSUPPORTED) > - unsupported = 1; > + unsupported = true; > else { > if (reg_azf & AZF_AC97_REG_REAL_IO_WRITE) > snd_azf3328_mixer_outw( > @@ -808,7 +808,7 @@ snd_azf3328_mixer_ac97_write(struct snd_ > */ > break; > default: > - unsupported = 1; > + unsupported = true; > break; > } > } > @@ -1559,7 +1559,7 @@ snd_azf3328_pcm_trigger(struct snd_pcm_s > struct snd_azf3328_codec_data *codec = runtime->private_data; > int result = 0; > u16 flags1; > - bool previously_muted = 0; > + bool previously_muted = false; > bool is_main_mixer_playback_codec = (AZF_CODEC_PLAYBACK == codec->type); > > snd_azf3328_dbgcalls("snd_azf3328_pcm_trigger cmd %d\n", cmd); >

1 0

[alsa-devel] [PATCH v2] ALSA : hda - not use assigned converters for all unused pins
by mengdong.lin＠intel.com 26 Sep '13

26 Sep '13

From: Mengdong Lin <mengdong.lin(a)intel.com> BIOS can mark a pin as "no physical connection" if the port is used by an integrated display which is not audio capable. And audio driver will overlook such pins. On Haswell, such a disconneted pin will keep muted and connected to the 1st converter by default. But if the 1st convertor is assigned to a connected pin for audio streaming. The muted disconnected pin can make the connected pin no sound output. So this patch avoids using assigned converters for all unused pins for Haswell, including the disconected pins. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 3d8cd044..7ea0245 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1149,32 +1149,43 @@ static int hdmi_choose_cvt(struct hda_codec *codec, } static void haswell_config_cvts(struct hda_codec *codec, - int pin_id, int mux_id) + hda_nid_t pin_nid, int mux_idx) { struct hdmi_spec *spec = codec->spec; - struct hdmi_spec_per_pin *per_pin; - int pin_idx, mux_idx; - int curr; - int err; + hda_nid_t nid, end_nid; + int cvt_idx, curr; + struct hdmi_spec_per_cvt *per_cvt; - for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { - per_pin = get_pin(spec, pin_idx); + /* configure all pins, including "no physical connection" ones */ + end_nid = codec->start_nid + codec->num_nodes; + for (nid = codec->start_nid; nid < end_nid; nid++) { + unsigned int wid_caps = get_wcaps(codec, nid); + unsigned int wid_type = get_wcaps_type(wid_caps); - if (pin_idx == pin_id) + if (wid_type != AC_WID_PIN) continue; - curr = snd_hda_codec_read(codec, per_pin->pin_nid, 0, + if (nid == pin_nid) + continue; + + curr = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0); + if (curr != mux_idx) + continue; - /* Choose another unused converter */ - if (curr == mux_id) { - err = hdmi_choose_cvt(codec, pin_idx, NULL, &mux_idx); - if (err < 0) - return; - snd_printdd("HDMI: choose converter %d for pin %d\n", mux_idx, pin_idx); - snd_hda_codec_write_cache(codec, per_pin->pin_nid, 0, + /* choose an unassigned converter. The conveters in the + * connection list are in the same order as in the codec. + */ + for (cvt_idx = 0; cvt_idx < spec->num_cvts; cvt_idx++) { + per_cvt = get_cvt(spec, cvt_idx); + if (!per_cvt->assigned) { + snd_printdd("choose cvt %d for pin nid %d\n", + cvt_idx, nid); + snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, - mux_idx); + cvt_idx); + break; + } } } } @@ -1216,7 +1227,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, /* configure unused pins to choose other converters */ if (is_haswell(codec)) - haswell_config_cvts(codec, pin_idx, mux_idx); + haswell_config_cvts(codec, per_pin->pin_nid, mux_idx); snd_hda_spdif_ctls_assign(codec, pin_idx, per_cvt->cvt_nid); -- 1.8.1.2

3 2

[alsa-devel] [PATCH] ASoC: sn95031: Convert to regmap
by Mark Brown 25 Sep '13

25 Sep '13

From: Mark Brown <broonie(a)linaro.org> This moves us towards being able to remove the duplicated register I/O functionality in ASoC. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/sn95031.c | 35 ++++++++++++++++++++--------------- 1 file changed, 20 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index dba26e63..13045f2 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -164,30 +164,28 @@ static unsigned int sn95031_get_mic_bias(struct snd_soc_codec *codec) } /*end - adc helper functions */ -static inline unsigned int sn95031_read(struct snd_soc_codec *codec, - unsigned int reg) +static int sn95031_read(void *ctx, unsigned int reg, unsigned int *val) { u8 value = 0; int ret; ret = intel_scu_ipc_ioread8(reg, &value); - if (ret) - pr_err("read of %x failed, err %d\n", reg, ret); - return value; + if (ret == 0) + *val = value; + return ret; } -static inline int sn95031_write(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value) +static int sn95031_write(void *ctx, unsigned int reg, unsigned int value) { - int ret; - - ret = intel_scu_ipc_iowrite8(reg, value); - if (ret) - pr_err("write of %x failed, err %d\n", reg, ret); - return ret; + return intel_scu_ipc_iowrite8(reg, value); } +static const struct regmap_config sn95031_regmap = { + .reg_read = sn95031_read, + .reg_write = sn95031_write, +}; + static int sn95031_set_vaud_bias(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -827,6 +825,8 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec) { pr_debug("codec_probe called\n"); + snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + /* PCM interface config * This sets the pcm rx slot conguration to max 6 slots * for max 4 dais (2 stereo and 2 mono) @@ -886,8 +886,6 @@ static int sn95031_codec_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver sn95031_codec = { .probe = sn95031_codec_probe, .remove = sn95031_codec_remove, - .read = sn95031_read, - .write = sn95031_write, .set_bias_level = sn95031_set_vaud_bias, .idle_bias_off = true, .dapm_widgets = sn95031_dapm_widgets, @@ -898,7 +896,14 @@ static struct snd_soc_codec_driver sn95031_codec = { static int sn95031_device_probe(struct platform_device *pdev) { + struct regmap *regmap; + pr_debug("codec device probe called for %s\n", dev_name(&pdev->dev)); + + regmap = devm_regmap_init(&pdev->dev, NULL, NULL, &sn95031_regmap); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + return snd_soc_register_codec(&pdev->dev, &sn95031_codec, sn95031_dais, ARRAY_SIZE(sn95031_dais)); } -- 1.8.4.rc3

1 0

Re: [alsa-devel] IMX WM8740 I2S support
by Fabio Estevam 25 Sep '13

25 Sep '13

[Please always Cc the mailing list] On Wed, Sep 25, 2013 at 2:05 PM, stéphane cerveau <scerveau(a)gmail.com> wrote: > I'm using an old kernel from a BSP provided by a retailer. This list is about mainline kernel. If you would like to use mainline kernel we could help you here. If you decide to stay with the vendor BSP, then this is not the appropriate place for support.

2 1

[alsa-devel] IMX WM8740 I2S support
by stéphane cerveau 25 Sep '13

25 Sep '13

Dear all, I'm currently working on supporting WM8740 device software configured with imx6q system. For now my driver seems to be configured correctly. The playback is launched but i get: ALSA sound/core/pcm_lib.c:1822 playback write error (DMA or IRQ trouble? write error: Input/output error The imx ssi is not sending me any data on I2S bus( nothing can be seen on the oscillo). I would like to know how do I have to configure my master clock in order to launch the I2S. My master clock is for now 24MHz. Sorry to be confused a bit but i'm quite newbie with this device. Best regards. Stéphane

2 8

[alsa-devel] [PATCH] ASoC: ab8500: Explicitly set I/O up
by Mark Brown 25 Sep '13

25 Sep '13

From: Mark Brown <broonie(a)linaro.org> We do some I/O in probe so we need to ensure the I/O operations are fully set up then. Reported-by: Olof Johansson <olof(a)lixom.net> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/ab8500-codec.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index ca500fd..a0394a8 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2473,6 +2473,8 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec) dev_dbg(dev, "%s: Enter.\n", __func__); + snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + /* Setup AB8500 according to board-settings */ pdata = dev_get_platdata(dev->parent); -- 1.8.4.rc3

3 7

[alsa-devel] [PATCH] ASoC: imx-sgtl5000: do not use devres on a foreign device
by Philipp Zabel 25 Sep '13

25 Sep '13

Calling devm_clk_get with any device pointer other than our own confuses devres. Use clk_get instead. This avoids hitting the following warning in the imx-sgtl5000 error path: imx-sgtl5000 sound.12: snd_soc_register_card failed (-517) platform sound.12: Driver imx-sgtl5000 requests probe deferral ------------[ cut here ]------------ WARNING: CPU: 0 PID: 75 at drivers/base/dd.c:272 driver_probe_device+0x194/0x218() Modules linked in: snd_soc_sgtl5000(+) snd_soc_imx_sgtl5000 coda snd_soc_imx_audmux imx_sdma snd_soc_fsl_spdif snd_soc_fsl_ssi CPU: 0 PID: 75 Comm: udevd Not tainted 3.11.0-rc6+ #4682 Backtrace: [<80010bc4>] (dump_backtrace+0x0/0x10c) from [<80010d60>] (show_stack+0x18/0x1c) r6:00000110 r5:00000009 r4:00000000 r3:00000000 [<80010d48>] (show_stack+0x0/0x1c) from [<804f0764>] (dump_stack+0x20/0x28) [<804f0744>] (dump_stack+0x0/0x28) from [<8001a4a4>] (warn_slowpath_common+0x6c/0x8c) [<8001a438>] (warn_slowpath_common+0x0/0x8c) from [<8001a4e8>] (warn_slowpath_null+0x24/0x2c) r8:7f032000 r7:7f02f93c r6:cf8eaa54 r5:cf8eaa20 r4:80728a0c [<8001a4c4>] (warn_slowpath_null+0x0/0x2c) from [<80286bdc>] (driver_probe_device+0x194/0x218) [<80286a48>] (driver_probe_device+0x0/0x218) from [<80286cf4>] (__driver_attach+0x94/0x98) r7:00000000 r6:cf8eaa54 r5:7f02f93c r4:cf8eaa20 [<80286c60>] (__driver_attach+0x0/0x98) from [<802851c8>] (bus_for_each_dev+0x5c/0x90) r6:80286c60 r5:7f02f93c r4:00000000 r3:cf8ef03c [<8028516c>] (bus_for_each_dev+0x0/0x90) from [<80286654>] (driver_attach+0x24/0x28) r6:806d0424 r5:cf16a580 r4:7f02f93c [<80286630>] (driver_attach+0x0/0x28) from [<802861e4>] (bus_add_driver+0xdc/0x234) [<80286108>] (bus_add_driver+0x0/0x234) from [<802871d4>] (driver_register+0x80/0x154) r8:7f032000 r7:00000001 r6:7f02fa68 r5:7f02fa74 r4:7f02f93c [<80287154>] (driver_register+0x0/0x154) from [<8033c278>] (i2c_register_driver+0x34/0xbc) [<8033c244>] (i2c_register_driver+0x0/0xbc) from [<7f032018>] (sgtl5000_i2c_driver_init+0x18/0x24 [snd_soc_sgtl5000]) r5:7f02fa74 r4:cfb7ff48 [<7f032000>] (sgtl5000_i2c_driver_init+0x0/0x24 [snd_soc_sgtl5000]) from [<80008738>] (do_one_initcall+0xf4/0x150) [<80008644>] (do_one_initcall+0x0/0x150) from [<80053f64>] (load_module+0x174c/0x1db4) [<80052818>] (load_module+0x0/0x1db4) from [<800546ac>] (SyS_init_module+0xe0/0xf4) [<800545cc>] (SyS_init_module+0x0/0xf4) from [<8000e540>] (ret_fast_syscall+0x0/0x30) r6:00005b22 r5:00afed68 r4:00000000 ---[ end trace b24c5c3bb145dbdd ]--- Signed-off-by: Philipp Zabel <p.zabel(a)pengutronix.de> --- sound/soc/fsl/imx-sgtl5000.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 46c5b4f..52df7d5 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -128,7 +128,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) goto fail; } - data->codec_clk = devm_clk_get(&codec_dev->dev, NULL); + data->codec_clk = clk_get(&codec_dev->dev, NULL); if (IS_ERR(data->codec_clk)) { ret = PTR_ERR(data->codec_clk); goto fail; @@ -172,6 +172,8 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) return 0; fail: + if (!IS_ERR(data->codec_clk)) + clk_put(data->codec_clk); if (ssi_np) of_node_put(ssi_np); if (codec_np) @@ -185,6 +187,7 @@ static int imx_sgtl5000_remove(struct platform_device *pdev) struct imx_sgtl5000_data *data = platform_get_drvdata(pdev); snd_soc_unregister_card(&data->card); + clk_put(data->codec_clk); return 0; } -- 1.8.4.rc3

3 5

[alsa-devel] [PATCH 1/3] mfd: mc13xxx: Don't require lock for simple register I/O
by Mark Brown 25 Sep '13

25 Sep '13

From: Mark Brown <broonie(a)linaro.org> Since the conversion to regmap there has been no need for device level locking for I/O as regmap provides locking so remove the locks. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- drivers/mfd/mc13xxx-core.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/drivers/mfd/mc13xxx-core.c b/drivers/mfd/mc13xxx-core.c index 2a9b100..dbbf8ee 100644 --- a/drivers/mfd/mc13xxx-core.c +++ b/drivers/mfd/mc13xxx-core.c @@ -158,8 +158,6 @@ int mc13xxx_reg_read(struct mc13xxx *mc13xxx, unsigned int offset, u32 *val) { int ret; - BUG_ON(!mutex_is_locked(&mc13xxx->lock)); - if (offset > MC13XXX_NUMREGS) return -EINVAL; @@ -172,8 +170,6 @@ EXPORT_SYMBOL(mc13xxx_reg_read); int mc13xxx_reg_write(struct mc13xxx *mc13xxx, unsigned int offset, u32 val) { - BUG_ON(!mutex_is_locked(&mc13xxx->lock)); - dev_vdbg(mc13xxx->dev, "[0x%02x] <- 0x%06x\n", offset, val); if (offset > MC13XXX_NUMREGS || val > 0xffffff) @@ -186,7 +182,6 @@ EXPORT_SYMBOL(mc13xxx_reg_write); int mc13xxx_reg_rmw(struct mc13xxx *mc13xxx, unsigned int offset, u32 mask, u32 val) { - BUG_ON(!mutex_is_locked(&mc13xxx->lock)); BUG_ON(val & ~mask); dev_vdbg(mc13xxx->dev, "[0x%02x] <- 0x%06x (mask: 0x%06x)\n", offset, val, mask); -- 1.8.4.rc3

2 11

[alsa-devel] [PATCH 1/3] ASoC: tlv320aic26: Convert to table based control init
by Mark Brown 25 Sep '13

25 Sep '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/tlv320aic26.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 7b8f3d9..32c6b07 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -377,7 +377,7 @@ static DEVICE_ATTR(keyclick, 0644, aic26_keyclick_show, aic26_keyclick_set); static int aic26_probe(struct snd_soc_codec *codec) { struct aic26 *aic26 = dev_get_drvdata(codec->dev); - int ret, err, i, reg; + int ret, i, reg; aic26->codec = codec; @@ -403,12 +403,6 @@ static int aic26_probe(struct snd_soc_codec *codec) if (ret) dev_info(codec->dev, "error creating sysfs files\n"); - /* register controls */ - dev_dbg(codec->dev, "Registering controls\n"); - err = snd_soc_add_codec_controls(codec, aic26_snd_controls, - ARRAY_SIZE(aic26_snd_controls)); - WARN_ON(err < 0); - return 0; } @@ -418,6 +412,8 @@ static struct snd_soc_codec_driver aic26_soc_codec_dev = { .write = aic26_reg_write, .reg_cache_size = AIC26_NUM_REGS, .reg_word_size = sizeof(u16), + .controls = aic26_snd_controls, + .num_controls = ARRAY_SIZE(aic26_snd_controls), .dapm_widgets = tlv320aic26_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tlv320aic26_dapm_widgets), .dapm_routes = tlv320aic26_dapm_routes, -- 1.8.4.rc3

1 2

[alsa-devel] [PATCH 1/2] ASoC: tlv320aic32x4: Move GPIO acquisition to I2C probe
by Mark Brown 25 Sep '13

25 Sep '13

From: Mark Brown <broonie(a)linaro.org> This is more idiomatic and interacts better with deferred probe. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/tlv320aic32x4.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 2ed57d4..cf70bf8 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -617,16 +617,11 @@ static int aic32x4_probe(struct snd_soc_codec *codec) { struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); u32 tmp_reg; - int ret; codec->hw_write = (hw_write_t) i2c_master_send; codec->control_data = aic32x4->control_data; if (aic32x4->rstn_gpio >= 0) { - ret = devm_gpio_request_one(codec->dev, aic32x4->rstn_gpio, - GPIOF_OUT_INIT_LOW, "tlv320aic32x4 rstn"); - if (ret != 0) - return ret; ndelay(10); gpio_set_value(aic32x4->rstn_gpio, 1); } @@ -735,6 +730,13 @@ static int aic32x4_i2c_probe(struct i2c_client *i2c, aic32x4->rstn_gpio = -1; } + if (aic32x4->rstn_gpio >= 0) { + ret = devm_gpio_request_one(&i2c->dev, aic32x4->rstn_gpio, + GPIOF_OUT_INIT_LOW, "tlv320aic32x4 rstn"); + if (ret != 0) + return ret; + } + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_aic32x4, &aic32x4_dai, 1); return ret; -- 1.8.4.rc3

1 1

[alsa-devel] [TINYCOMPRESS][PATCH 1/1] compress: Must check for POLLERR before POLLOUT/POLLIN
by Richard Fitzgerald 25 Sep '13

25 Sep '13

In the case of error the ALSA compressed driver sets revents as (POLLOUT | POLLWRNORM | POLLERR) or (POLLIN | POLLWRNORM | POLLERR). So we can't assume that POLLOUT or POLLIN indicate success, we must check for POLLERR first. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- compress.c | 18 +++++++++--------- 1 files changed, 9 insertions(+), 9 deletions(-) diff --git a/compress.c b/compress.c index 0e71c28..5cd0966 100644 --- a/compress.c +++ b/compress.c @@ -378,6 +378,9 @@ int compress_write(struct compress *compress, const void *buf, unsigned int size return total; ret = poll(&fds, 1, compress->max_poll_wait_ms); + if (fds.revents & POLLERR) { + return oops(compress, EIO, "poll returned error!"); + } /* A pause will cause -EBADFD or zero. * This is not an error, just stop writing */ if ((ret == 0) || (ret == -EBADFD)) @@ -387,9 +390,6 @@ int compress_write(struct compress *compress, const void *buf, unsigned int size if (fds.revents & POLLOUT) { continue; } - if (fds.revents & POLLERR) { - return oops(compress, EIO, "poll returned error!"); - } } /* write avail bytes */ if (size > avail.avail) @@ -438,6 +438,9 @@ int compress_read(struct compress *compress, void *buf, unsigned int size) return total; ret = poll(&fds, 1, compress->max_poll_wait_ms); + if (fds.revents & POLLERR) { + return oops(compress, EIO, "poll returned error!"); + } /* A pause will cause -EBADFD or zero. * This is not an error, just stop reading */ if ((ret == 0) || (ret == -EBADFD)) @@ -447,9 +450,6 @@ int compress_read(struct compress *compress, void *buf, unsigned int size) if (fds.revents & POLLIN) { continue; } - if (fds.revents & POLLERR) { - return oops(compress, EIO, "poll returned error!"); - } } /* read avail bytes */ if (size > avail.avail) @@ -616,15 +616,15 @@ int compress_wait(struct compress *compress, int timeout_ms) fds.events = POLLOUT | POLLIN; ret = poll(&fds, 1, timeout_ms); + if (fds.revents & POLLERR) { + return oops(compress, EIO, "poll returned error!"); + } /* A pause will cause -EBADFD or zero. */ if ((ret < 0) && (ret != -EBADFD)) return oops(compress, errno, "poll error"); if (fds.revents & (POLLOUT | POLLIN)) { return 0; } - if (fds.revents & POLLERR) { - return oops(compress, EIO, "poll returned error!"); - } return ret; } -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ASoC: kirkwood: simplify clock handling
by Uwe Kleine-König 24 Sep '13

24 Sep '13

There is no need to not use extclk if it is identical to the main clk. The main motivation for this patch is dropping devm_clk_put which is used in a wrong way by all other users. While at it also extend the comments. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- Hello, there might be some further optimisations possible. I only note them here because I don't have the hardware to test: - only enable extclk if it a clock rate used that makes use of the external clock. Not sure if that works; hardware docs reading necessary. - only provide extclk if it's != the internal clock. - The code uses: priv->clk = devm_clk_get(&pdev->dev, np ? "internal" : NULL); i.e. provides a con_id in the dt-case. I think that using NULL unconditionally should also work, i.e. return the first clk associated to the device. OTOH the current code might make things clearer because it's more explicit. Best regards Uwe --- sound/soc/kirkwood/kirkwood-i2s.c | 24 +++++++++++++++--------- 1 file changed, 15 insertions(+), 9 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 0f3d73d..8224f93 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -104,16 +104,25 @@ static void kirkwood_set_rate(struct snd_soc_dai *dai, uint32_t clks_ctrl; if (rate == 44100 || rate == 48000 || rate == 96000) { - /* use internal dco for the supported rates - * defined in kirkwood_i2s_dai */ + /* + * use internal dco for the supported rates + * defined in kirkwood_i2s_dai + * Note: For these specific rates the dco is also used if + * kirkwood_i2s_dai_extclk is in use. + */ dev_dbg(dai->dev, "%s: dco set rate = %lu\n", __func__, rate); kirkwood_set_dco(priv->io, rate); clks_ctrl = KIRKWOOD_MCLK_SOURCE_DCO; } else { - /* use the external clock for the other rates - * defined in kirkwood_i2s_dai_extclk */ + /* + * use the external clock for the other rates + * defined in kirkwood_i2s_dai_extclk + * This case isn't reached if kirkwood_i2s_dai is in use (and so + * extclk isn't valid) because it only supports the three rates + * above. + */ dev_dbg(dai->dev, "%s: extclk set rate = %lu -> %lu\n", __func__, rate, 256 * rate); clk_set_rate(priv->extclk, 256 * rate); @@ -497,12 +506,9 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) priv->extclk = devm_clk_get(&pdev->dev, "extclk"); if (!IS_ERR(priv->extclk)) { - if (priv->extclk == priv->clk) { - devm_clk_put(&pdev->dev, priv->extclk); - priv->extclk = ERR_PTR(-EINVAL); - } else { + clk_prepare_enable(priv->extclk); + if (priv->extclk != priv->clk) { dev_info(&pdev->dev, "found external clock\n"); - clk_prepare_enable(priv->extclk); soc_dai = &kirkwood_i2s_dai_extclk; } } -- 1.8.4.rc3

3 4

Re: [alsa-devel] [PATCH v2 4/4] ASoC: rsnd: use regmap instead of original register mapping method
by Mark Brown 24 Sep '13

24 Sep '13

On Tue, Sep 17, 2013 at 10:23:58PM +0900, Kuninori Morimoto wrote: > For example, SSI device have many ports (0-8), > but ADG is only 1 device. > On this driver, all devices are using rsnd_write/read() to > access register. OK, this sounds like the module should have this information - ie check something in the module structure rather than asking regmap.

2 4

[alsa-devel] [PATCH 1/3] ASoC: tlv320aic23: Remove #defines for I2C
by Mark Brown 24 Sep '13

24 Sep '13

From: Mark Brown <broonie(a)linaro.org> The only control interface supported by this driver is I2C so there is no need for conditional compilation around the control interface. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/tlv320aic23.c | 25 +------------------------ 1 file changed, 1 insertion(+), 24 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 31762eb..3299459 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -613,7 +613,6 @@ static struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = { .num_dapm_routes = ARRAY_SIZE(tlv320aic23_intercon), }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) /* * If the i2c layer weren't so broken, we could pass this kind of data * around @@ -660,29 +659,7 @@ static struct i2c_driver tlv320aic23_i2c_driver = { .id_table = tlv320aic23_id, }; -#endif - -static int __init tlv320aic23_modinit(void) -{ - int ret; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - ret = i2c_add_driver(&tlv320aic23_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register TLV320AIC23 I2C driver: %d\n", - ret); - } -#endif - return ret; -} -module_init(tlv320aic23_modinit); - -static void __exit tlv320aic23_exit(void) -{ -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - i2c_del_driver(&tlv320aic23_i2c_driver); -#endif -} -module_exit(tlv320aic23_exit); +module_i2c_driver(tlv320aic23_i2c_driver); MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver"); MODULE_AUTHOR("Arun KS <arunks(a)mistralsolutions.com>"); -- 1.8.4.rc3

1 2

[alsa-devel] [PATCH v3 1/2] ASoC: kirkwood: clk: probe defer when clock not yet ready
by Jean-Francois Moine 24 Sep '13

24 Sep '13

At probe time, a clock device may not be ready when some other device wants to use it. This patch lets the functions clk_get/devm_clk_get return a probe defer when the clock is defined in the DT but not yet available. It also fixes an erroneous call to clk_get_sys() when __clk_get() fails. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- v2: fix __clk_get() failure (from Russell King) Note: the associated patch v2 2/2 is unchanged and not resent --- drivers/clk/clk.c | 2 +- drivers/clk/clkdev.c | 7 ++++++- 2 files changed, 7 insertions(+), 2 deletions(-) diff --git a/drivers/clk/clk.c b/drivers/clk/clk.c index a004769..582abc8 100644 --- a/drivers/clk/clk.c +++ b/drivers/clk/clk.c @@ -2181,7 +2181,7 @@ EXPORT_SYMBOL_GPL(of_clk_del_provider); struct clk *of_clk_get_from_provider(struct of_phandle_args *clkspec) { struct of_clk_provider *provider; - struct clk *clk = ERR_PTR(-ENOENT); + struct clk *clk = ERR_PTR(-EPROBE_DEFER); /* Check if we have such a provider in our array */ mutex_lock(&of_clk_lock); diff --git a/drivers/clk/clkdev.c b/drivers/clk/clkdev.c index 442a313..1f402b9 100644 --- a/drivers/clk/clkdev.c +++ b/drivers/clk/clkdev.c @@ -157,7 +157,12 @@ struct clk *clk_get(struct device *dev, const char *con_id) if (dev) { clk = of_clk_get_by_name(dev->of_node, con_id); - if (!IS_ERR(clk) && __clk_get(clk)) + if (!IS_ERR(clk)) { + if (!__clk_get(clk)) + clk = ERR_PTR(-ENOENT); + return clk; + } + if (PTR_ERR(clk) == -EPROBE_DEFER) return clk; } -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

1 0

[alsa-devel] "ASoC: ab8500: Convert register I/O to regmap" breaks snowball
by Olof Johansson 24 Sep '13

24 Sep '13

Mark, Lee, [I wasn't subscribed to alsa-devel when the patch was posted, so I'm sending this as a standalone email instead of as a reply to that] linux-next 20130924 fails to boot on snowball (u8500_defconfig, non-DT boot): Unable to handle kernel NULL pointer dereference at virtual address 00000000 pgd = c0004000 [00000000] *pgd=00000000 Internal error: Oops: 80000005 [#1] PREEMPT SMP ARM Modules linked in: CPU: 0 PID: 1 Comm: swapper/0 Not tainted 3.12.0-rc1-00003-gff795d6 #135 task: ef060000 ti: ef064000 task.ti: ef064000 PC is at 0x0 LR is at snd_soc_update_bits+0x4c/0x88 pc : [<00000000>] lr : [<c02d4f98>] psr: 60000013 sp : ef065d48 ip : 00000020 fp : ef0d054c r10: 00000000 r9 : ef3c0810 r8 : 00000000 r7 : 00000014 r6 : 00000000 r5 : 00000007 r4 : ef36ea00 r3 : 00000000 r2 : 00000007 r1 : 00000014 r0 : ef36ea00 Flags: nZCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment kernel Control: 10c5787d Table: 0000404a DAC: 00000015 Process swapper/0 (pid: 1, stack limit = 0xef064240) Stack: (0xef065d48 to 0xef066000) 5d40: ef153410 c0550bb8 ef153410 ef36ea00 c053dc64 ef36ea00 5d60: c053dc64 c03d5e0c ef36eaac c02e4144 00000000 02000000 ef36ea00 00000000 5d80: c055a688 ef17bc10 ef0a7b00 c02e4358 00000000 00000000 00000000 ef36ea00 5da0: c055a5dc c055a688 c055daa4 c055aa0c ef36eaac c02d74fc c03d592c c055a678 5dc0: ee8115c8 c055daa4 ef0d00c0 00000000 00000000 c055a668 c03d5bf4 c02d8258 5de0: c055dab4 00000000 00000004 eec2d990 c055db7c ef0d0780 c055dbac c055db9c 5e00: c055a6a8 ef36ea30 ef36ea00 ef0d087c ef0d084c ef36ea30 c04a98f4 c055dac4 5e20: c021ce54 c055daa4 000005b8 c053f7f0 c055da64 c051dde8 00000069 ef064028 5e40: 00000000 c02d8ef0 00000000 c055da50 00000000 c02e684c c05924fc c053f7f0 5e60: 00000000 c055da64 c051dde8 c021bc88 c021bc70 c021aa94 00000000 c053f7f0 5e80: c055da64 c053f824 00000000 c021ac80 00000000 c055da64 c021abf4 c0218f6c 5ea0: ef03265c ef1b6734 c055da64 ef192500 c054fea8 c021a220 c0475770 c055da64 5ec0: 00000006 c055da64 00000006 c0563c80 c0563c80 c021b25c 00000000 c052c5f4 5ee0: 00000006 c0008798 ef010900 c04787e4 ef09c100 c03ac5e0 00000000 00000001 5f00: 000014dc c0540b9c 60000013 00000001 00000003 00000000 c0da19bd c03bf6b0 5f20: 00000069 c0036aa4 c04de34c 00000006 c0da19c8 00000006 00000000 c052c5f4 5f40: 00000006 c0563c80 c0563c80 c05014dc 00000069 c0523fec c0523fe0 c0501bc4 5f60: 00000006 00000006 c05014dc 00000000 88000000 00000000 00000000 00000000 5f80: c03a14ec 00000000 00000000 00000000 00000000 00000000 00000000 c03a14f4 5fa0: 00000000 00000000 c03a14ec c000e438 00000000 00000000 00000000 00000000 5fc0: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 5fe0: 00000000 00000000 00000000 00000000 00000013 00000000 00000000 20000000 [<c02d4f98>] (snd_soc_update_bits+0x4c/0x88) from [<c02e4144>] (ab8500_audio_setup_mics+0xc8/0x118) [<c02e4144>] (ab8500_audio_setup_mics+0xc8/0x118) from [<c02e4358>] (ab8500_codec_probe+0x180/0x374) [<c02e4358>] (ab8500_codec_probe+0x180/0x374) from [<c02d74fc>] (soc_probe_codec+0x178/0x2f4) [<c02d74fc>] (soc_probe_codec+0x178/0x2f4) from [<c02d8258>] (snd_soc_instantiate_card+0x794/0x11c0) [<c02d8258>] (snd_soc_instantiate_card+0x794/0x11c0) from [<c02d8ef0>] (snd_soc_register_card+0x26c/0x324) [<c02d8ef0>] (snd_soc_register_card+0x26c/0x324) from [<c02e684c>] (mop500_probe+0xd0/0x118) [<c02e684c>] (mop500_probe+0xd0/0x118) from [<c021bc88>] (platform_drv_probe+0x18/0x1c) [<c021bc88>] (platform_drv_probe+0x18/0x1c) from [<c021aa94>] (driver_probe_device+0x10c/0x228) ... I bisected it down to: commit ff795d614bfa62a3c6fc0bcb75cb8842e5a87892 Author: Mark Brown <broonie(a)linaro.org> Date: Fri Sep 20 10:38:16 2013 +0100 ASoC: ab8500: Convert register I/O to regmap As part of a general push to eliminate the duplicated register I/O support in ASoC convert ab8500 to use regmap. Signed-off-by: Mark Brown <broonie(a)linaro.org> Acked-by: Lee Jones <lee.jones(a)linaro.org> ... but I didn't look closer at what was actually broken. -Olof

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[alsa-devel] [PATCH 1/3] ASoC: max98095: Remove custom hw_write() implementation
by Mark Brown 24 Sep '13

24 Sep '13

From: Mark Brown <broonie(a)linaro.org> The registers that are being kept uncached are marked as volatile anyway so the call has no practical impact. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/max98095.c | 25 ++++--------------------- 1 file changed, 4 insertions(+), 21 deletions(-) diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 8dbcacd..29c266b 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -612,23 +612,6 @@ static int max98095_volatile(struct snd_soc_codec *codec, unsigned int reg) } /* - * Filter coefficients are in a separate register segment - * and they share the address space of the normal registers. - * The coefficient registers do not need or share the cache. - */ -static int max98095_hw_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - int ret; - - codec->cache_bypass = 1; - ret = snd_soc_write(codec, reg, value); - codec->cache_bypass = 0; - - return ret ? -EIO : 0; -} - -/* * Load equalizer DSP coefficient configurations registers */ static void m98095_eq_band(struct snd_soc_codec *codec, unsigned int dai, @@ -648,8 +631,8 @@ static void m98095_eq_band(struct snd_soc_codec *codec, unsigned int dai, /* Step through the registers and coefs */ for (i = 0; i < M98095_COEFS_PER_BAND; i++) { - max98095_hw_write(codec, eq_reg++, M98095_BYTE1(coefs[i])); - max98095_hw_write(codec, eq_reg++, M98095_BYTE0(coefs[i])); + snd_soc_write(codec, eq_reg++, M98095_BYTE1(coefs[i])); + snd_soc_write(codec, eq_reg++, M98095_BYTE0(coefs[i])); } } @@ -673,8 +656,8 @@ static void m98095_biquad_band(struct snd_soc_codec *codec, unsigned int dai, /* Step through the registers and coefs */ for (i = 0; i < M98095_COEFS_PER_BAND; i++) { - max98095_hw_write(codec, bq_reg++, M98095_BYTE1(coefs[i])); - max98095_hw_write(codec, bq_reg++, M98095_BYTE0(coefs[i])); + snd_soc_write(codec, bq_reg++, M98095_BYTE1(coefs[i])); + snd_soc_write(codec, bq_reg++, M98095_BYTE0(coefs[i])); } } -- 1.8.4.rc3

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[alsa-devel] [PATCH 1/5] ASoC: Docs: Update codec documentation
by Liam Girdwood 24 Sep '13

24 Sep '13

Update the codec class driver documentation and bring it up to date with the current code base. This includes API changes, regmap and multi component. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- Documentation/sound/alsa/soc/codec.txt | 46 +++++++++++----------------------- 1 file changed, 15 insertions(+), 31 deletions(-) diff --git a/Documentation/sound/alsa/soc/codec.txt b/Documentation/sound/alsa/soc/codec.txt index bce23a4..db5f9c9 100644 --- a/Documentation/sound/alsa/soc/codec.txt +++ b/Documentation/sound/alsa/soc/codec.txt @@ -1,22 +1,23 @@ -ASoC Codec Driver -================= +ASoC Codec Class Driver +======================= -The codec driver is generic and hardware independent code that configures the -codec to provide audio capture and playback. It should contain no code that is -specific to the target platform or machine. All platform and machine specific -code should be added to the platform and machine drivers respectively. +The codec class driver is generic and hardware independent code that configures +the codec, FM, MODEM, BT or external DSP to provide audio capture and playback. +It should contain no code that is specific to the target platform or machine. +All platform and machine specific code should be added to the platform and +machine drivers respectively. -Each codec driver *must* provide the following features:- +Each codec class driver *must* provide the following features:- 1) Codec DAI and PCM configuration - 2) Codec control IO - using I2C, 3 Wire(SPI) or both APIs + 2) Codec control IO - using RegMap API 3) Mixers and audio controls 4) Codec audio operations + 5) DAPM description. + 6) DAPM event handler. Optionally, codec drivers can also provide:- - 5) DAPM description. - 6) DAPM event handler. 7) DAC Digital mute control. Its probably best to use this guide in conjunction with the existing codec @@ -64,26 +65,9 @@ struct snd_soc_dai_driver wm8731_dai = { 2 - Codec control IO -------------------- The codec can usually be controlled via an I2C or SPI style interface -(AC97 combines control with data in the DAI). The codec drivers provide -functions to read and write the codec registers along with supplying a -register cache:- - - /* IO control data and register cache */ - void *control_data; /* codec control (i2c/3wire) data */ - void *reg_cache; - -Codec read/write should do any data formatting and call the hardware -read write below to perform the IO. These functions are called by the -core and ALSA when performing DAPM or changing the mixer:- - - unsigned int (*read)(struct snd_soc_codec *, unsigned int); - int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); - -Codec hardware IO functions - usually points to either the I2C, SPI or AC97 -read/write:- - - hw_write_t hw_write; - hw_read_t hw_read; +(AC97 combines control with data in the DAI). The codec driver should use the +Regmap API for all codec IO. Please see include/linux/regmap.h and existing +codec drivers for example regmap usage. 3 - Mixers and audio controls @@ -127,7 +111,7 @@ Defines a stereo enumerated control 4 - Codec Audio Operations -------------------------- -The codec driver also supports the following ALSA operations:- +The codec driver also supports the following ALSA PCM operations:- /* SoC audio ops */ struct snd_soc_ops { -- 1.8.1.2

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[alsa-devel] [PATCH] ASoC: fsl: fsl_ssi: Fix simultaneous capture and playback
by Fabio Estevam 24 Sep '13

24 Sep '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> When doing simultaneous capture and playback on a mx6 board we get the following error: $ arecord -f cd | aplay -f cd imx-sgtl5000 sound.13: set sample size in capture stream first fsl-ssi-dai 2028000.ssi: ASoC: can't open interface 2028000.ssi: -11 ALSA lib pcm_dmix.c:1018:(snd_pcm_dmix_ open) unable to open slave aplay: main:660: audio open error: Device or resource busy Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo The 'arecord -f cd | aplay -f cd' always trigger cause the 'if (!first_runtime->sample_bits)' block to be true which returns an error. Adjust the logic inside fsl_ssi_startup(), so that we do not always hit the error when playing 'arecord | aplay' line for the first time. Reported-by: Chris Clepper <cgclepper(a)gmail.com> Suggested-by: Nicolin Chen <b42378(a)freescale.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 15 ++++----------- 1 file changed, 4 insertions(+), 11 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 6ac8730..cdbb641 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -469,19 +469,12 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, * parameters, then the second stream may be * constrained to the wrong sample rate or size. */ - if (!first_runtime->sample_bits) { - dev_err(substream->pcm->card->dev, - "set sample size in %s stream first\n", - substream->stream == - SNDRV_PCM_STREAM_PLAYBACK - ? "capture" : "playback"); - return -EAGAIN; - } - - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + if (first_runtime->sample_bits) { + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, first_runtime->sample_bits, first_runtime->sample_bits); + } } ssi_private->second_stream = substream; -- 1.8.1.2

2 1

[alsa-devel] Audio Miniconference 2013 - Edinburgh 21st October
by Mark Brown 24 Sep '13

24 Sep '13

As in previous years we're planning to have an audio mini-summit to bring everyone together to discuss the current state of audio in Linux, try to resolve outstanding design issues and do some hacking. This year we will be meeting in Edinburgh on 21st October as part of the Linux Foundation conferences being held that week. What topics would people like to discuss? There are a number of ongoing issues like DSP integration and related topics such as compressed audio and I know that there are some issues which have come up while people have been working on deploying UCM especially with PulseAudio integration. What other areas are of interest to people? To help with planning could you please fill out the form at: https://docs.google.com/forms/d/105uav9tkwwkDU4P1T3IOWjfsa3kgtDbFAJOlHgBQR5… if you're planning to attend, ideally by 21st September. Since the facilities are being provided by the Linux Foundation attendees will need to be registered for at least one of the other conferences that week, if that's an issue please let us know. There's a space for topics of interest on the form but please also bring things up on the list so other people can get a feel for what's going on and ideally so we can resolve them all via e-mail!

2 2

[alsa-devel] [PATCH 0/2][RFC] snd_soc_codec include snd_soc_component
by Kuninori Morimoto 24 Sep '13

24 Sep '13

Hi Mark, Lars These are very rough patch for merging snd_soc_codec <-> snd_soc_component. As Lars idea, "codec" includes "component" on this patch, but it doesn't care about "platform" here, since "platform" doesn't have "dai". Current component is caring "dai", but... ? (and what about "card" ?) Anyway, this patch focus only codec <-> component. The "driver code" was not big difference, since current component is very small at this point. Attached 1st patch is rough version of ASoC core code. the main topic on this patch is that snd_soc_register_dai[s] will be called via component. But, we will need many "codec dai name exchange", becouse 1) the dai name rule on 1-dai / multi-dai are different, it was depend on snd_soc_register_dai[s]. 1) component is using both snd_soc_register_dai[s] inside (this was for "cpu" side limit) 2) codec have been used snd_soc_register_dais even though it was not multi-dai 2nd patch is rough version of "code dai name exchange" for it. Of course we need more and more codec driver patches, but it is wm8978 only as trial now. I guess "codec driver" should have "component driver" pointer too ? (it has .name only at this point, and 2nd patch includes it) We can move some feature from codec to component after this step ? In my quick check, below codec have 0 dai (= we need care it on component ?) sound/soc/codecs/lm4857.c sound/soc/codecs/max9768.c sound/soc/codecs/max9877.c sound/soc/codecs/wm2000.c sound/soc/codecs/wm9090.c below codecs have 1 dai (= we need name exchange on platform) sound/soc/codecs/ac97.c sound/soc/codecs/ad1836.c sound/soc/codecs/ad193x.c sound/soc/codecs/ad193x.c sound/soc/codecs/ad1980.c sound/soc/codecs/ad73311.c sound/soc/codecs/adau1701.c sound/soc/codecs/ads117x.c sound/soc/codecs/ak4104.c sound/soc/codecs/ak4535.c sound/soc/codecs/ak4554.c sound/soc/codecs/ak4642.c sound/soc/codecs/ak4671.c sound/soc/codecs/ak5386.c sound/soc/codecs/alc5623.c sound/soc/codecs/alc5632.c sound/soc/codecs/bt-sco.c sound/soc/codecs/cq93vc.c sound/soc/codecs/cs4270.c sound/soc/codecs/cs4271.c sound/soc/codecs/cs4271.c sound/soc/codecs/cs42l51.c sound/soc/codecs/cs42l52.c sound/soc/codecs/cx20442.c sound/soc/codecs/da7210.c sound/soc/codecs/da7210.c sound/soc/codecs/da7213.c sound/soc/codecs/da9055.c sound/soc/codecs/dmic.c sound/soc/codecs/hdmi.c sound/soc/codecs/jz4740.c sound/soc/codecs/max9850.c sound/soc/codecs/ml26124.c sound/soc/codecs/pcm1681.c sound/soc/codecs/pcm1792a.c sound/soc/codecs/pcm3008.c sound/soc/codecs/sgtl5000.c sound/soc/codecs/si476x.c sound/soc/codecs/spdif_receiver.c sound/soc/codecs/spdif_transmitter.c sound/soc/codecs/ssm2518.c sound/soc/codecs/ssm2602.c sound/soc/codecs/ssm2602.c sound/soc/codecs/sta32x.c sound/soc/codecs/sta529.c sound/soc/codecs/tas5086.c sound/soc/codecs/tlv320aic23.c sound/soc/codecs/tlv320aic26.c sound/soc/codecs/tlv320aic32x4.c sound/soc/codecs/tlv320aic3x.c sound/soc/codecs/tlv320dac33.c sound/soc/codecs/uda134x.c sound/soc/codecs/wl1273.c sound/soc/codecs/wm1250-ev1.c sound/soc/codecs/wm2200.c sound/soc/codecs/wm8350.c sound/soc/codecs/wm8400.c sound/soc/codecs/wm8510.c sound/soc/codecs/wm8510.c sound/soc/codecs/wm8523.c sound/soc/codecs/wm8711.c sound/soc/codecs/wm8711.c sound/soc/codecs/wm8727.c sound/soc/codecs/wm8728.c sound/soc/codecs/wm8728.c sound/soc/codecs/wm8731.c sound/soc/codecs/wm8731.c sound/soc/codecs/wm8737.c sound/soc/codecs/wm8737.c sound/soc/codecs/wm8741.c sound/soc/codecs/wm8741.c sound/soc/codecs/wm8750.c sound/soc/codecs/wm8750.c sound/soc/codecs/wm8770.c sound/soc/codecs/wm8782.c sound/soc/codecs/wm8804.c sound/soc/codecs/wm8804.c sound/soc/codecs/wm8900.c sound/soc/codecs/wm8900.c sound/soc/codecs/wm8903.c sound/soc/codecs/wm8904.c sound/soc/codecs/wm8940.c sound/soc/codecs/wm8955.c sound/soc/codecs/wm8960.c sound/soc/codecs/wm8961.c sound/soc/codecs/wm8962.c sound/soc/codecs/wm8971.c sound/soc/codecs/wm8974.c sound/soc/codecs/wm8978.c sound/soc/codecs/wm8983.c sound/soc/codecs/wm8983.c sound/soc/codecs/wm8985.c sound/soc/codecs/wm8985.c sound/soc/codecs/wm8988.c sound/soc/codecs/wm8988.c sound/soc/codecs/wm8990.c sound/soc/codecs/wm8991.c sound/soc/codecs/wm8993.c sound/soc/codecs/wm9081.c sound/soc/codecs/max98090.c sound/soc/codecs/rt5631.c below codecs have multi dai sound/soc/codecs/88pm860x-codec.c sound/soc/codecs/ab8500-codec.c sound/soc/codecs/adau1373.c sound/soc/codecs/adav80x.c sound/soc/codecs/ak4641.c sound/soc/codecs/cs42l73.c sound/soc/codecs/da732x.c sound/soc/codecs/isabelle.c sound/soc/codecs/lm49453.c sound/soc/codecs/max98088.c sound/soc/codecs/max98095.c sound/soc/codecs/rt5640.c sound/soc/codecs/sn95031.c sound/soc/codecs/stac9766.c sound/soc/codecs/twl4030.c sound/soc/codecs/twl6040.c sound/soc/codecs/uda1380.c sound/soc/codecs/wm0010.c sound/soc/codecs/wm5100.c sound/soc/codecs/wm5102.c sound/soc/codecs/wm5110.c sound/soc/codecs/wm8580.c sound/soc/codecs/wm8753.c sound/soc/codecs/wm8776.c sound/soc/codecs/wm8994.c sound/soc/codecs/wm8995.c sound/soc/codecs/wm8996.c sound/soc/codecs/wm8997.c sound/soc/codecs/wm9705.c sound/soc/codecs/wm9712.c sound/soc/codecs/wm9713.c and, below codec has both 1-dai, and multi-dai sound/soc/codecs/mc13783.c Kuninori Morimoto (2): [RFC] ASoC: snd_soc_codec include snd_soc_component [RFC] ASoC: wm8978: use component arch/arm/mach-shmobile/board-armadillo800eva.c | 2 +- include/sound/soc.h | 33 ++++++++++-------- sound/soc/codecs/wm8978.c | 6 ++++ sound/soc/soc-core.c | 44 ++++++++++++++++++------ 4 files changed, 59 insertions(+), 26 deletions(-)

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Re: [alsa-devel] [PATCH 3/3] ASoC: max98088: Convert to direct regmap API usage
by Dylan Reid 24 Sep '13

24 Sep '13

On Mon, Sep 23, 2013 at 9:58 AM, Mark Brown <broonie(a)kernel.org> wrote: > From: Mark Brown <broonie(a)linaro.org> > > This saves code and moves us towards removing the redundant register I/O > implementation in ASoC. > > Signed-off-by: Mark Brown <broonie(a)linaro.org> all three successfully tested. > --- > sound/soc/codecs/max98088.c | 580 +++++++++++++++++++------------------------- > 1 file changed, 251 insertions(+), 329 deletions(-) > > diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c > index 8896d5e..31912d5 100644 > --- a/sound/soc/codecs/max98088.c > +++ b/sound/soc/codecs/max98088.c > @@ -15,6 +15,7 @@ > #include <linux/delay.h> > #include <linux/pm.h> > #include <linux/i2c.h> > +#include <linux/regmap.h> > #include <sound/core.h> > #include <sound/pcm.h> > #include <sound/pcm_params.h> > @@ -38,6 +39,7 @@ struct max98088_cdata { > }; > > struct max98088_priv { > + struct regmap *regmap; > enum max98088_type devtype; > struct max98088_pdata *pdata; > unsigned int sysclk; > @@ -54,278 +56,206 @@ struct max98088_priv { > unsigned int extmic_mode; > }; > > -static const u8 max98088_reg[M98088_REG_CNT] = { > - 0x00, /* 00 IRQ status */ > - 0x00, /* 01 MIC status */ > - 0x00, /* 02 jack status */ > - 0x00, /* 03 battery voltage */ > - 0x00, /* 04 */ > - 0x00, /* 05 */ > - 0x00, /* 06 */ > - 0x00, /* 07 */ > - 0x00, /* 08 */ > - 0x00, /* 09 */ > - 0x00, /* 0A */ > - 0x00, /* 0B */ > - 0x00, /* 0C */ > - 0x00, /* 0D */ > - 0x00, /* 0E */ > - 0x00, /* 0F interrupt enable */ > - > - 0x00, /* 10 master clock */ > - 0x00, /* 11 DAI1 clock mode */ > - 0x00, /* 12 DAI1 clock control */ > - 0x00, /* 13 DAI1 clock control */ > - 0x00, /* 14 DAI1 format */ > - 0x00, /* 15 DAI1 clock */ > - 0x00, /* 16 DAI1 config */ > - 0x00, /* 17 DAI1 TDM */ > - 0x00, /* 18 DAI1 filters */ > - 0x00, /* 19 DAI2 clock mode */ > - 0x00, /* 1A DAI2 clock control */ > - 0x00, /* 1B DAI2 clock control */ > - 0x00, /* 1C DAI2 format */ > - 0x00, /* 1D DAI2 clock */ > - 0x00, /* 1E DAI2 config */ > - 0x00, /* 1F DAI2 TDM */ > - > - 0x00, /* 20 DAI2 filters */ > - 0x00, /* 21 data config */ > - 0x00, /* 22 DAC mixer */ > - 0x00, /* 23 left ADC mixer */ > - 0x00, /* 24 right ADC mixer */ > - 0x00, /* 25 left HP mixer */ > - 0x00, /* 26 right HP mixer */ > - 0x00, /* 27 HP control */ > - 0x00, /* 28 left REC mixer */ > - 0x00, /* 29 right REC mixer */ > - 0x00, /* 2A REC control */ > - 0x00, /* 2B left SPK mixer */ > - 0x00, /* 2C right SPK mixer */ > - 0x00, /* 2D SPK control */ > - 0x00, /* 2E sidetone */ > - 0x00, /* 2F DAI1 playback level */ > - > - 0x00, /* 30 DAI1 playback level */ > - 0x00, /* 31 DAI2 playback level */ > - 0x00, /* 32 DAI2 playbakc level */ > - 0x00, /* 33 left ADC level */ > - 0x00, /* 34 right ADC level */ > - 0x00, /* 35 MIC1 level */ > - 0x00, /* 36 MIC2 level */ > - 0x00, /* 37 INA level */ > - 0x00, /* 38 INB level */ > - 0x00, /* 39 left HP volume */ > - 0x00, /* 3A right HP volume */ > - 0x00, /* 3B left REC volume */ > - 0x00, /* 3C right REC volume */ > - 0x00, /* 3D left SPK volume */ > - 0x00, /* 3E right SPK volume */ > - 0x00, /* 3F MIC config */ > - > - 0x00, /* 40 MIC threshold */ > - 0x00, /* 41 excursion limiter filter */ > - 0x00, /* 42 excursion limiter threshold */ > - 0x00, /* 43 ALC */ > - 0x00, /* 44 power limiter threshold */ > - 0x00, /* 45 power limiter config */ > - 0x00, /* 46 distortion limiter config */ > - 0x00, /* 47 audio input */ > - 0x00, /* 48 microphone */ > - 0x00, /* 49 level control */ > - 0x00, /* 4A bypass switches */ > - 0x00, /* 4B jack detect */ > - 0x00, /* 4C input enable */ > - 0x00, /* 4D output enable */ > - 0xF0, /* 4E bias control */ > - 0x00, /* 4F DAC power */ > - > - 0x0F, /* 50 DAC power */ > - 0x00, /* 51 system */ > - 0x00, /* 52 DAI1 EQ1 */ > - 0x00, /* 53 DAI1 EQ1 */ > - 0x00, /* 54 DAI1 EQ1 */ > - 0x00, /* 55 DAI1 EQ1 */ > - 0x00, /* 56 DAI1 EQ1 */ > - 0x00, /* 57 DAI1 EQ1 */ > - 0x00, /* 58 DAI1 EQ1 */ > - 0x00, /* 59 DAI1 EQ1 */ > - 0x00, /* 5A DAI1 EQ1 */ > - 0x00, /* 5B DAI1 EQ1 */ > - 0x00, /* 5C DAI1 EQ2 */ > - 0x00, /* 5D DAI1 EQ2 */ > - 0x00, /* 5E DAI1 EQ2 */ > - 0x00, /* 5F DAI1 EQ2 */ > - > - 0x00, /* 60 DAI1 EQ2 */ > - 0x00, /* 61 DAI1 EQ2 */ > - 0x00, /* 62 DAI1 EQ2 */ > - 0x00, /* 63 DAI1 EQ2 */ > - 0x00, /* 64 DAI1 EQ2 */ > - 0x00, /* 65 DAI1 EQ2 */ > - 0x00, /* 66 DAI1 EQ3 */ > - 0x00, /* 67 DAI1 EQ3 */ > - 0x00, /* 68 DAI1 EQ3 */ > - 0x00, /* 69 DAI1 EQ3 */ > - 0x00, /* 6A DAI1 EQ3 */ > - 0x00, /* 6B DAI1 EQ3 */ > - 0x00, /* 6C DAI1 EQ3 */ > - 0x00, /* 6D DAI1 EQ3 */ > - 0x00, /* 6E DAI1 EQ3 */ > - 0x00, /* 6F DAI1 EQ3 */ > - > - 0x00, /* 70 DAI1 EQ4 */ > - 0x00, /* 71 DAI1 EQ4 */ > - 0x00, /* 72 DAI1 EQ4 */ > - 0x00, /* 73 DAI1 EQ4 */ > - 0x00, /* 74 DAI1 EQ4 */ > - 0x00, /* 75 DAI1 EQ4 */ > - 0x00, /* 76 DAI1 EQ4 */ > - 0x00, /* 77 DAI1 EQ4 */ > - 0x00, /* 78 DAI1 EQ4 */ > - 0x00, /* 79 DAI1 EQ4 */ > - 0x00, /* 7A DAI1 EQ5 */ > - 0x00, /* 7B DAI1 EQ5 */ > - 0x00, /* 7C DAI1 EQ5 */ > - 0x00, /* 7D DAI1 EQ5 */ > - 0x00, /* 7E DAI1 EQ5 */ > - 0x00, /* 7F DAI1 EQ5 */ > - > - 0x00, /* 80 DAI1 EQ5 */ > - 0x00, /* 81 DAI1 EQ5 */ > - 0x00, /* 82 DAI1 EQ5 */ > - 0x00, /* 83 DAI1 EQ5 */ > - 0x00, /* 84 DAI2 EQ1 */ > - 0x00, /* 85 DAI2 EQ1 */ > - 0x00, /* 86 DAI2 EQ1 */ > - 0x00, /* 87 DAI2 EQ1 */ > - 0x00, /* 88 DAI2 EQ1 */ > - 0x00, /* 89 DAI2 EQ1 */ > - 0x00, /* 8A DAI2 EQ1 */ > - 0x00, /* 8B DAI2 EQ1 */ > - 0x00, /* 8C DAI2 EQ1 */ > - 0x00, /* 8D DAI2 EQ1 */ > - 0x00, /* 8E DAI2 EQ2 */ > - 0x00, /* 8F DAI2 EQ2 */ > - > - 0x00, /* 90 DAI2 EQ2 */ > - 0x00, /* 91 DAI2 EQ2 */ > - 0x00, /* 92 DAI2 EQ2 */ > - 0x00, /* 93 DAI2 EQ2 */ > - 0x00, /* 94 DAI2 EQ2 */ > - 0x00, /* 95 DAI2 EQ2 */ > - 0x00, /* 96 DAI2 EQ2 */ > - 0x00, /* 97 DAI2 EQ2 */ > - 0x00, /* 98 DAI2 EQ3 */ > - 0x00, /* 99 DAI2 EQ3 */ > - 0x00, /* 9A DAI2 EQ3 */ > - 0x00, /* 9B DAI2 EQ3 */ > - 0x00, /* 9C DAI2 EQ3 */ > - 0x00, /* 9D DAI2 EQ3 */ > - 0x00, /* 9E DAI2 EQ3 */ > - 0x00, /* 9F DAI2 EQ3 */ > - > - 0x00, /* A0 DAI2 EQ3 */ > - 0x00, /* A1 DAI2 EQ3 */ > - 0x00, /* A2 DAI2 EQ4 */ > - 0x00, /* A3 DAI2 EQ4 */ > - 0x00, /* A4 DAI2 EQ4 */ > - 0x00, /* A5 DAI2 EQ4 */ > - 0x00, /* A6 DAI2 EQ4 */ > - 0x00, /* A7 DAI2 EQ4 */ > - 0x00, /* A8 DAI2 EQ4 */ > - 0x00, /* A9 DAI2 EQ4 */ > - 0x00, /* AA DAI2 EQ4 */ > - 0x00, /* AB DAI2 EQ4 */ > - 0x00, /* AC DAI2 EQ5 */ > - 0x00, /* AD DAI2 EQ5 */ > - 0x00, /* AE DAI2 EQ5 */ > - 0x00, /* AF DAI2 EQ5 */ > - > - 0x00, /* B0 DAI2 EQ5 */ > - 0x00, /* B1 DAI2 EQ5 */ > - 0x00, /* B2 DAI2 EQ5 */ > - 0x00, /* B3 DAI2 EQ5 */ > - 0x00, /* B4 DAI2 EQ5 */ > - 0x00, /* B5 DAI2 EQ5 */ > - 0x00, /* B6 DAI1 biquad */ > - 0x00, /* B7 DAI1 biquad */ > - 0x00, /* B8 DAI1 biquad */ > - 0x00, /* B9 DAI1 biquad */ > - 0x00, /* BA DAI1 biquad */ > - 0x00, /* BB DAI1 biquad */ > - 0x00, /* BC DAI1 biquad */ > - 0x00, /* BD DAI1 biquad */ > - 0x00, /* BE DAI1 biquad */ > - 0x00, /* BF DAI1 biquad */ > - > - 0x00, /* C0 DAI2 biquad */ > - 0x00, /* C1 DAI2 biquad */ > - 0x00, /* C2 DAI2 biquad */ > - 0x00, /* C3 DAI2 biquad */ > - 0x00, /* C4 DAI2 biquad */ > - 0x00, /* C5 DAI2 biquad */ > - 0x00, /* C6 DAI2 biquad */ > - 0x00, /* C7 DAI2 biquad */ > - 0x00, /* C8 DAI2 biquad */ > - 0x00, /* C9 DAI2 biquad */ > - 0x00, /* CA */ > - 0x00, /* CB */ > - 0x00, /* CC */ > - 0x00, /* CD */ > - 0x00, /* CE */ > - 0x00, /* CF */ > - > - 0x00, /* D0 */ > - 0x00, /* D1 */ > - 0x00, /* D2 */ > - 0x00, /* D3 */ > - 0x00, /* D4 */ > - 0x00, /* D5 */ > - 0x00, /* D6 */ > - 0x00, /* D7 */ > - 0x00, /* D8 */ > - 0x00, /* D9 */ > - 0x00, /* DA */ > - 0x70, /* DB */ > - 0x00, /* DC */ > - 0x00, /* DD */ > - 0x00, /* DE */ > - 0x00, /* DF */ > - > - 0x00, /* E0 */ > - 0x00, /* E1 */ > - 0x00, /* E2 */ > - 0x00, /* E3 */ > - 0x00, /* E4 */ > - 0x00, /* E5 */ > - 0x00, /* E6 */ > - 0x00, /* E7 */ > - 0x00, /* E8 */ > - 0x00, /* E9 */ > - 0x00, /* EA */ > - 0x00, /* EB */ > - 0x00, /* EC */ > - 0x00, /* ED */ > - 0x00, /* EE */ > - 0x00, /* EF */ > - > - 0x00, /* F0 */ > - 0x00, /* F1 */ > - 0x00, /* F2 */ > - 0x00, /* F3 */ > - 0x00, /* F4 */ > - 0x00, /* F5 */ > - 0x00, /* F6 */ > - 0x00, /* F7 */ > - 0x00, /* F8 */ > - 0x00, /* F9 */ > - 0x00, /* FA */ > - 0x00, /* FB */ > - 0x00, /* FC */ > - 0x00, /* FD */ > - 0x00, /* FE */ > - 0x00, /* FF */ > +static const struct reg_default max98088_reg[] = { > + { 0xf, 0x00 }, /* 0F interrupt enable */ > + > + { 0x10, 0x00 }, /* 10 master clock */ > + { 0x11, 0x00 }, /* 11 DAI1 clock mode */ > + { 0x12, 0x00 }, /* 12 DAI1 clock control */ > + { 0x13, 0x00 }, /* 13 DAI1 clock control */ > + { 0x14, 0x00 }, /* 14 DAI1 format */ > + { 0x15, 0x00 }, /* 15 DAI1 clock */ > + { 0x16, 0x00 }, /* 16 DAI1 config */ > + { 0x17, 0x00 }, /* 17 DAI1 TDM */ > + { 0x18, 0x00 }, /* 18 DAI1 filters */ > + { 0x19, 0x00 }, /* 19 DAI2 clock mode */ > + { 0x1a, 0x00 }, /* 1A DAI2 clock control */ > + { 0x1b, 0x00 }, /* 1B DAI2 clock control */ > + { 0x1c, 0x00 }, /* 1C DAI2 format */ > + { 0x1d, 0x00 }, /* 1D DAI2 clock */ > + { 0x1e, 0x00 }, /* 1E DAI2 config */ > + { 0x1f, 0x00 }, /* 1F DAI2 TDM */ > + > + { 0x20, 0x00 }, /* 20 DAI2 filters */ > + { 0x21, 0x00 }, /* 21 data config */ > + { 0x22, 0x00 }, /* 22 DAC mixer */ > + { 0x23, 0x00 }, /* 23 left ADC mixer */ > + { 0x24, 0x00 }, /* 24 right ADC mixer */ > + { 0x25, 0x00 }, /* 25 left HP mixer */ > + { 0x26, 0x00 }, /* 26 right HP mixer */ > + { 0x27, 0x00 }, /* 27 HP control */ > + { 0x28, 0x00 }, /* 28 left REC mixer */ > + { 0x29, 0x00 }, /* 29 right REC mixer */ > + { 0x2a, 0x00 }, /* 2A REC control */ > + { 0x2b, 0x00 }, /* 2B left SPK mixer */ > + { 0x2c, 0x00 }, /* 2C right SPK mixer */ > + { 0x2d, 0x00 }, /* 2D SPK control */ > + { 0x2e, 0x00 }, /* 2E sidetone */ > + { 0x2f, 0x00 }, /* 2F DAI1 playback level */ > + > + { 0x30, 0x00 }, /* 30 DAI1 playback level */ > + { 0x31, 0x00 }, /* 31 DAI2 playback level */ > + { 0x32, 0x00 }, /* 32 DAI2 playbakc level */ > + { 0x33, 0x00 }, /* 33 left ADC level */ > + { 0x34, 0x00 }, /* 34 right ADC level */ > + { 0x35, 0x00 }, /* 35 MIC1 level */ > + { 0x36, 0x00 }, /* 36 MIC2 level */ > + { 0x37, 0x00 }, /* 37 INA level */ > + { 0x38, 0x00 }, /* 38 INB level */ > + { 0x39, 0x00 }, /* 39 left HP volume */ > + { 0x3a, 0x00 }, /* 3A right HP volume */ > + { 0x3b, 0x00 }, /* 3B left REC volume */ > + { 0x3c, 0x00 }, /* 3C right REC volume */ > + { 0x3d, 0x00 }, /* 3D left SPK volume */ > + { 0x3e, 0x00 }, /* 3E right SPK volume */ > + { 0x3f, 0x00 }, /* 3F MIC config */ > + > + { 0x40, 0x00 }, /* 40 MIC threshold */ > + { 0x41, 0x00 }, /* 41 excursion limiter filter */ > + { 0x42, 0x00 }, /* 42 excursion limiter threshold */ > + { 0x43, 0x00 }, /* 43 ALC */ > + { 0x44, 0x00 }, /* 44 power limiter threshold */ > + { 0x45, 0x00 }, /* 45 power limiter config */ > + { 0x46, 0x00 }, /* 46 distortion limiter config */ > + { 0x47, 0x00 }, /* 47 audio input */ > + { 0x48, 0x00 }, /* 48 microphone */ > + { 0x49, 0x00 }, /* 49 level control */ > + { 0x4a, 0x00 }, /* 4A bypass switches */ > + { 0x4b, 0x00 }, /* 4B jack detect */ > + { 0x4c, 0x00 }, /* 4C input enable */ > + { 0x4d, 0x00 }, /* 4D output enable */ > + { 0x4e, 0xF0 }, /* 4E bias control */ > + { 0x4f, 0x00 }, /* 4F DAC power */ > + > + { 0x50, 0x0F }, /* 50 DAC power */ > + { 0x51, 0x00 }, /* 51 system */ > + { 0x52, 0x00 }, /* 52 DAI1 EQ1 */ > + { 0x53, 0x00 }, /* 53 DAI1 EQ1 */ > + { 0x54, 0x00 }, /* 54 DAI1 EQ1 */ > + { 0x55, 0x00 }, /* 55 DAI1 EQ1 */ > + { 0x56, 0x00 }, /* 56 DAI1 EQ1 */ > + { 0x57, 0x00 }, /* 57 DAI1 EQ1 */ > + { 0x58, 0x00 }, /* 58 DAI1 EQ1 */ > + { 0x59, 0x00 }, /* 59 DAI1 EQ1 */ > + { 0x5a, 0x00 }, /* 5A DAI1 EQ1 */ > + { 0x5b, 0x00 }, /* 5B DAI1 EQ1 */ > + { 0x5c, 0x00 }, /* 5C DAI1 EQ2 */ > + { 0x5d, 0x00 }, /* 5D DAI1 EQ2 */ > + { 0x5e, 0x00 }, /* 5E DAI1 EQ2 */ > + { 0x5f, 0x00 }, /* 5F DAI1 EQ2 */ > + > + { 0x60, 0x00 }, /* 60 DAI1 EQ2 */ > + { 0x61, 0x00 }, /* 61 DAI1 EQ2 */ > + { 0x62, 0x00 }, /* 62 DAI1 EQ2 */ > + { 0x63, 0x00 }, /* 63 DAI1 EQ2 */ > + { 0x64, 0x00 }, /* 64 DAI1 EQ2 */ > + { 0x65, 0x00 }, /* 65 DAI1 EQ2 */ > + { 0x66, 0x00 }, /* 66 DAI1 EQ3 */ > + { 0x67, 0x00 }, /* 67 DAI1 EQ3 */ > + { 0x68, 0x00 }, /* 68 DAI1 EQ3 */ > + { 0x69, 0x00 }, /* 69 DAI1 EQ3 */ > + { 0x6a, 0x00 }, /* 6A DAI1 EQ3 */ > + { 0x6b, 0x00 }, /* 6B DAI1 EQ3 */ > + { 0x6c, 0x00 }, /* 6C DAI1 EQ3 */ > + { 0x6d, 0x00 }, /* 6D DAI1 EQ3 */ > + { 0x6e, 0x00 }, /* 6E DAI1 EQ3 */ > + { 0x6f, 0x00 }, /* 6F DAI1 EQ3 */ > + > + { 0x70, 0x00 }, /* 70 DAI1 EQ4 */ > + { 0x71, 0x00 }, /* 71 DAI1 EQ4 */ > + { 0x72, 0x00 }, /* 72 DAI1 EQ4 */ > + { 0x73, 0x00 }, /* 73 DAI1 EQ4 */ > + { 0x74, 0x00 }, /* 74 DAI1 EQ4 */ > + { 0x75, 0x00 }, /* 75 DAI1 EQ4 */ > + { 0x76, 0x00 }, /* 76 DAI1 EQ4 */ > + { 0x77, 0x00 }, /* 77 DAI1 EQ4 */ > + { 0x78, 0x00 }, /* 78 DAI1 EQ4 */ > + { 0x79, 0x00 }, /* 79 DAI1 EQ4 */ > + { 0x7a, 0x00 }, /* 7A DAI1 EQ5 */ > + { 0x7b, 0x00 }, /* 7B DAI1 EQ5 */ > + { 0x7c, 0x00 }, /* 7C DAI1 EQ5 */ > + { 0x7d, 0x00 }, /* 7D DAI1 EQ5 */ > + { 0x7e, 0x00 }, /* 7E DAI1 EQ5 */ > + { 0x7f, 0x00 }, /* 7F DAI1 EQ5 */ > + > + { 0x80, 0x00 }, /* 80 DAI1 EQ5 */ > + { 0x81, 0x00 }, /* 81 DAI1 EQ5 */ > + { 0x82, 0x00 }, /* 82 DAI1 EQ5 */ > + { 0x83, 0x00 }, /* 83 DAI1 EQ5 */ > + { 0x84, 0x00 }, /* 84 DAI2 EQ1 */ > + { 0x85, 0x00 }, /* 85 DAI2 EQ1 */ > + { 0x86, 0x00 }, /* 86 DAI2 EQ1 */ > + { 0x87, 0x00 }, /* 87 DAI2 EQ1 */ > + { 0x88, 0x00 }, /* 88 DAI2 EQ1 */ > + { 0x89, 0x00 }, /* 89 DAI2 EQ1 */ > + { 0x8a, 0x00 }, /* 8A DAI2 EQ1 */ > + { 0x8b, 0x00 }, /* 8B DAI2 EQ1 */ > + { 0x8c, 0x00 }, /* 8C DAI2 EQ1 */ > + { 0x8d, 0x00 }, /* 8D DAI2 EQ1 */ > + { 0x8e, 0x00 }, /* 8E DAI2 EQ2 */ > + { 0x8f, 0x00 }, /* 8F DAI2 EQ2 */ > + > + { 0x90, 0x00 }, /* 90 DAI2 EQ2 */ > + { 0x91, 0x00 }, /* 91 DAI2 EQ2 */ > + { 0x92, 0x00 }, /* 92 DAI2 EQ2 */ > + { 0x93, 0x00 }, /* 93 DAI2 EQ2 */ > + { 0x94, 0x00 }, /* 94 DAI2 EQ2 */ > + { 0x95, 0x00 }, /* 95 DAI2 EQ2 */ > + { 0x96, 0x00 }, /* 96 DAI2 EQ2 */ > + { 0x97, 0x00 }, /* 97 DAI2 EQ2 */ > + { 0x98, 0x00 }, /* 98 DAI2 EQ3 */ > + { 0x99, 0x00 }, /* 99 DAI2 EQ3 */ > + { 0x9a, 0x00 }, /* 9A DAI2 EQ3 */ > + { 0x9b, 0x00 }, /* 9B DAI2 EQ3 */ > + { 0x9c, 0x00 }, /* 9C DAI2 EQ3 */ > + { 0x9d, 0x00 }, /* 9D DAI2 EQ3 */ > + { 0x9e, 0x00 }, /* 9E DAI2 EQ3 */ > + { 0x9f, 0x00 }, /* 9F DAI2 EQ3 */ > + > + { 0xa0, 0x00 }, /* A0 DAI2 EQ3 */ > + { 0xa1, 0x00 }, /* A1 DAI2 EQ3 */ > + { 0xa2, 0x00 }, /* A2 DAI2 EQ4 */ > + { 0xa3, 0x00 }, /* A3 DAI2 EQ4 */ > + { 0xa4, 0x00 }, /* A4 DAI2 EQ4 */ > + { 0xa5, 0x00 }, /* A5 DAI2 EQ4 */ > + { 0xa6, 0x00 }, /* A6 DAI2 EQ4 */ > + { 0xa7, 0x00 }, /* A7 DAI2 EQ4 */ > + { 0xa8, 0x00 }, /* A8 DAI2 EQ4 */ > + { 0xa9, 0x00 }, /* A9 DAI2 EQ4 */ > + { 0xaa, 0x00 }, /* AA DAI2 EQ4 */ > + { 0xab, 0x00 }, /* AB DAI2 EQ4 */ > + { 0xac, 0x00 }, /* AC DAI2 EQ5 */ > + { 0xad, 0x00 }, /* AD DAI2 EQ5 */ > + { 0xae, 0x00 }, /* AE DAI2 EQ5 */ > + { 0xaf, 0x00 }, /* AF DAI2 EQ5 */ > + > + { 0xb0, 0x00 }, /* B0 DAI2 EQ5 */ > + { 0xb1, 0x00 }, /* B1 DAI2 EQ5 */ > + { 0xb2, 0x00 }, /* B2 DAI2 EQ5 */ > + { 0xb3, 0x00 }, /* B3 DAI2 EQ5 */ > + { 0xb4, 0x00 }, /* B4 DAI2 EQ5 */ > + { 0xb5, 0x00 }, /* B5 DAI2 EQ5 */ > + { 0xb6, 0x00 }, /* B6 DAI1 biquad */ > + { 0xb7, 0x00 }, /* B7 DAI1 biquad */ > + { 0xb8 ,0x00 }, /* B8 DAI1 biquad */ > + { 0xb9, 0x00 }, /* B9 DAI1 biquad */ > + { 0xba, 0x00 }, /* BA DAI1 biquad */ > + { 0xbb, 0x00 }, /* BB DAI1 biquad */ > + { 0xbc, 0x00 }, /* BC DAI1 biquad */ > + { 0xbd, 0x00 }, /* BD DAI1 biquad */ > + { 0xbe, 0x00 }, /* BE DAI1 biquad */ > + { 0xbf, 0x00 }, /* BF DAI1 biquad */ > + > + { 0xc0, 0x00 }, /* C0 DAI2 biquad */ > + { 0xc1, 0x00 }, /* C1 DAI2 biquad */ > + { 0xc2, 0x00 }, /* C2 DAI2 biquad */ > + { 0xc3, 0x00 }, /* C3 DAI2 biquad */ > + { 0xc4, 0x00 }, /* C4 DAI2 biquad */ > + { 0xc5, 0x00 }, /* C5 DAI2 biquad */ > + { 0xc6, 0x00 }, /* C6 DAI2 biquad */ > + { 0xc7, 0x00 }, /* C7 DAI2 biquad */ > + { 0xc8, 0x00 }, /* C8 DAI2 biquad */ > + { 0xc9, 0x00 }, /* C9 DAI2 biquad */ > }; > > static struct { > @@ -606,11 +536,27 @@ static struct { > { 0xFF, 0x00, 1 }, /* FF */ > }; > > -static int max98088_volatile_register(struct snd_soc_codec *codec, unsigned int reg) > +static bool max98088_readable_register(struct device *dev, unsigned int reg) > +{ > + return max98088_access[reg].readable; > +} > + > +static bool max98088_volatile_register(struct device *dev, unsigned int reg) > { > return max98088_access[reg].vol; > } > > +static const struct regmap_config max98088_regmap = { > + .reg_bits = 8, > + .val_bits = 8, > + > + .readable_reg = max98088_readable_register, > + .volatile_reg = max98088_volatile_register, > + > + .reg_defaults = max98088_reg, > + .num_reg_defaults = ARRAY_SIZE(max98088_reg), > + .cache_type = REGCACHE_RBTREE, > +}; > > /* > * Load equalizer DSP coefficient configurations registers > @@ -1610,58 +1556,34 @@ static int max98088_dai2_digital_mute(struct snd_soc_dai *codec_dai, int mute) > return 0; > } > > -static void max98088_sync_cache(struct snd_soc_codec *codec) > -{ > - u8 *reg_cache = codec->reg_cache; > - int i; > - > - if (!codec->cache_sync) > - return; > - > - codec->cache_only = 0; > - > - /* write back cached values if they're writeable and > - * different from the hardware default. > - */ > - for (i = 1; i < codec->driver->reg_cache_size; i++) { > - if (!max98088_access[i].writable) > - continue; > - > - if (reg_cache[i] == max98088_reg[i]) > - continue; > - > - snd_soc_write(codec, i, reg_cache[i]); > - } > - > - codec->cache_sync = 0; > -} > - > static int max98088_set_bias_level(struct snd_soc_codec *codec, > enum snd_soc_bias_level level) > { > - switch (level) { > - case SND_SOC_BIAS_ON: > - break; > - > - case SND_SOC_BIAS_PREPARE: > - break; > - > - case SND_SOC_BIAS_STANDBY: > - if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) > - max98088_sync_cache(codec); > - > - snd_soc_update_bits(codec, M98088_REG_4C_PWR_EN_IN, > - M98088_MBEN, M98088_MBEN); > - break; > - > - case SND_SOC_BIAS_OFF: > - snd_soc_update_bits(codec, M98088_REG_4C_PWR_EN_IN, > - M98088_MBEN, 0); > - codec->cache_sync = 1; > - break; > - } > - codec->dapm.bias_level = level; > - return 0; > + struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); > + > + switch (level) { > + case SND_SOC_BIAS_ON: > + break; > + > + case SND_SOC_BIAS_PREPARE: > + break; > + > + case SND_SOC_BIAS_STANDBY: > + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) > + regcache_sync(max98088->regmap); > + > + snd_soc_update_bits(codec, M98088_REG_4C_PWR_EN_IN, > + M98088_MBEN, M98088_MBEN); > + break; > + > + case SND_SOC_BIAS_OFF: > + snd_soc_update_bits(codec, M98088_REG_4C_PWR_EN_IN, > + M98088_MBEN, 0); > + regcache_mark_dirty(max98088->regmap); > + break; > + } > + codec->dapm.bias_level = level; > + return 0; > } > > #define MAX98088_RATES SNDRV_PCM_RATE_8000_96000 > @@ -1988,9 +1910,9 @@ static int max98088_probe(struct snd_soc_codec *codec) > struct max98088_cdata *cdata; > int ret = 0; > > - codec->cache_sync = 1; > + regcache_mark_dirty(max98088->regmap); > > - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); > + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); > if (ret != 0) { > dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); > return ret; > @@ -2070,10 +1992,6 @@ static struct snd_soc_codec_driver soc_codec_dev_max98088 = { > .set_bias_level = max98088_set_bias_level, > .controls = max98088_snd_controls, > .num_controls = ARRAY_SIZE(max98088_snd_controls), > - .reg_cache_size = ARRAY_SIZE(max98088_reg), > - .reg_word_size = sizeof(u8), > - .reg_cache_default = max98088_reg, > - .volatile_register = max98088_volatile_register, > .dapm_widgets = max98088_dapm_widgets, > .num_dapm_widgets = ARRAY_SIZE(max98088_dapm_widgets), > .dapm_routes = max98088_audio_map, > @@ -2081,7 +1999,7 @@ static struct snd_soc_codec_driver soc_codec_dev_max98088 = { > }; > > static int max98088_i2c_probe(struct i2c_client *i2c, > - const struct i2c_device_id *id) > + const struct i2c_device_id *id) > { > struct max98088_priv *max98088; > int ret; > @@ -2091,6 +2009,10 @@ static int max98088_i2c_probe(struct i2c_client *i2c, > if (max98088 == NULL) > return -ENOMEM; > > + max98088->regmap = devm_regmap_init_i2c(i2c, &max98088_regmap); > + if (IS_ERR(max98088->regmap)) > + return PTR_ERR(max98088->regmap); > + > max98088->devtype = id->driver_data; > > i2c_set_clientdata(i2c, max98088); > -- > 1.8.4.rc3 >

1 0

[alsa-devel] [PATCH 1/6] ASoC: tlv320aic3x: Remove nonsense comment for register cache
by Mark Brown 24 Sep '13

24 Sep '13

From: Mark Brown <broonie(a)linaro.org> Every statement in this comment is incorrect either through bitrot or (mostly) through never having corresponded to reality in the first place. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/tlv320aic3x.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 6e3f269..3abbff3 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -90,12 +90,6 @@ struct aic3x_priv { enum aic3x_micbias_voltage micbias_vg; }; -/* - * AIC3X register cache - * We can't read the AIC3X register space when we are - * using 2 wire for device control, so we cache them instead. - * There is no point in caching the reset register - */ static const u8 aic3x_reg[AIC3X_CACHEREGNUM] = { 0x00, 0x00, 0x00, 0x10, /* 0 */ 0x04, 0x00, 0x00, 0x00, /* 4 */ -- 1.8.4.rc3

1 5

[alsa-devel] [PATCH] ASoC: max9850: Convert to direct regmap API usage
by Mark Brown 23 Sep '13

23 Sep '13

From: Mark Brown <broonie(a)linaro.org> This prepares for removal of the duplicated register I/O functionality in ASoC. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/max9850.c | 39 +++++++++++++++++++++++++++++---------- 1 file changed, 29 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c index 58c38a5..c5dd617 100644 --- a/sound/soc/codecs/max9850.c +++ b/sound/soc/codecs/max9850.c @@ -18,6 +18,7 @@ #include <linux/module.h> #include <linux/init.h> #include <linux/i2c.h> +#include <linux/regmap.h> #include <linux/slab.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -27,18 +28,26 @@ #include "max9850.h" struct max9850_priv { + struct regmap *regmap; unsigned int sysclk; }; /* max9850 register cache */ -static const u8 max9850_reg[MAX9850_CACHEREGNUM] = { - 0x00, 0x00, 0x0c, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00 +static const struct reg_default max9850_reg[] = { + { 2, 0x0c }, + { 3, 0x00 }, + { 4, 0x00 }, + { 5, 0x00 }, + { 6, 0x00 }, + { 7, 0x00 }, + { 8, 0x00 }, + { 9, 0x00 }, + { 10, 0x00 }, }; /* these registers are not used at the moment but provided for the sake of * completeness */ -static int max9850_volatile_register(struct snd_soc_codec *codec, - unsigned int reg) +static bool max9850_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { case MAX9850_STATUSA: @@ -49,6 +58,15 @@ static int max9850_volatile_register(struct snd_soc_codec *codec, } } +static const struct regmap_config max9850_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = MAX9850_DIGITAL_AUDIO, + .volatile_reg = max9850_volatile_register, + .cache_type = REGCACHE_RBTREE, +}; + static const unsigned int max9850_tlv[] = { TLV_DB_RANGE_HEAD(4), 0x18, 0x1f, TLV_DB_SCALE_ITEM(-7450, 400, 0), @@ -225,6 +243,7 @@ static int max9850_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) static int max9850_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct max9850_priv *max9850 = snd_soc_codec_get_drvdata(codec); int ret; switch (level) { @@ -234,7 +253,7 @@ static int max9850_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - ret = snd_soc_cache_sync(codec); + ret = regcache_sync(max9850->regmap); if (ret) { dev_err(codec->dev, "Failed to sync cache: %d\n", ret); @@ -295,7 +314,7 @@ static int max9850_probe(struct snd_soc_codec *codec) { int ret; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -316,10 +335,6 @@ static struct snd_soc_codec_driver soc_codec_dev_max9850 = { .suspend = max9850_suspend, .resume = max9850_resume, .set_bias_level = max9850_set_bias_level, - .reg_cache_size = ARRAY_SIZE(max9850_reg), - .reg_word_size = sizeof(u8), - .reg_cache_default = max9850_reg, - .volatile_register = max9850_volatile_register, .controls = max9850_controls, .num_controls = ARRAY_SIZE(max9850_controls), @@ -340,6 +355,10 @@ static int max9850_i2c_probe(struct i2c_client *i2c, if (max9850 == NULL) return -ENOMEM; + max9850->regmap = devm_regmap_init_i2c(i2c, &max9850_regmap); + if (IS_ERR(max9850->regmap)) + return PTR_ERR(max9850->regmap); + i2c_set_clientdata(i2c, max9850); ret = snd_soc_register_codec(&i2c->dev, -- 1.8.4.rc3

1 0

[alsa-devel] [PATCH 1/3] ASoC: max98088: Fix indentation
by Mark Brown 23 Sep '13

23 Sep '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/max98088.c | 46 ++++++++++++++++++++++----------------------- 1 file changed, 23 insertions(+), 23 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 566a367..391f669 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -38,20 +38,20 @@ struct max98088_cdata { }; struct max98088_priv { - enum max98088_type devtype; - struct max98088_pdata *pdata; - unsigned int sysclk; - struct max98088_cdata dai[2]; - int eq_textcnt; - const char **eq_texts; - struct soc_enum eq_enum; - u8 ina_state; - u8 inb_state; - unsigned int ex_mode; - unsigned int digmic; - unsigned int mic1pre; - unsigned int mic2pre; - unsigned int extmic_mode; + enum max98088_type devtype; + struct max98088_pdata *pdata; + unsigned int sysclk; + struct max98088_cdata dai[2]; + int eq_textcnt; + const char **eq_texts; + struct soc_enum eq_enum; + u8 ina_state; + u8 inb_state; + unsigned int ex_mode; + unsigned int digmic; + unsigned int mic1pre; + unsigned int mic2pre; + unsigned int extmic_mode; }; static const u8 max98088_reg[M98088_REG_CNT] = { @@ -2066,15 +2066,15 @@ static int max98088_remove(struct snd_soc_codec *codec) } static struct snd_soc_codec_driver soc_codec_dev_max98088 = { - .probe = max98088_probe, - .remove = max98088_remove, - .suspend = max98088_suspend, - .resume = max98088_resume, - .set_bias_level = max98088_set_bias_level, - .reg_cache_size = ARRAY_SIZE(max98088_reg), - .reg_word_size = sizeof(u8), - .reg_cache_default = max98088_reg, - .volatile_register = max98088_volatile_register, + .probe = max98088_probe, + .remove = max98088_remove, + .suspend = max98088_suspend, + .resume = max98088_resume, + .set_bias_level = max98088_set_bias_level, + .reg_cache_size = ARRAY_SIZE(max98088_reg), + .reg_word_size = sizeof(u8), + .reg_cache_default = max98088_reg, + .volatile_register = max98088_volatile_register, .dapm_widgets = max98088_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(max98088_dapm_widgets), .dapm_routes = max98088_audio_map, -- 1.8.4.rc3

1 2

Re: [alsa-devel] [PATCH 24/51] DMA-API: dma: pl330: add dma_set_mask_and_coherent() call
by Russell King - ARM Linux 23 Sep '13

23 Sep '13

On Fri, Sep 20, 2013 at 07:26:27PM +0200, Heiko Stübner wrote: > Am Donnerstag, 19. September 2013, 23:49:01 schrieb Russell King: > > The DMA API requires drivers to call the appropriate dma_set_mask() > > functions before doing any DMA mapping. Add this required call to > > the AMBA PL08x driver. > ^--- copy and paste error - should of course be PL330 Fixed, thanks.

2 1

[alsa-devel] fsl_ssi: Cannot do simultaneous capture and playback
by Fabio Estevam 23 Sep '13

23 Sep '13

Hi, I am trying to do simultaneous capture and playback on mx6 wandboard and I noticed that it only works if I run 'aplay first'. Here is the sequence that works: 1) aplay file.wav 2) Run 'alsamixer' and select line-in in the capture tab 3) Connect audio via line-in input 4) arecord -f cd -c 2 | aplay -f cd Then I am able to hear the playback. However, if I do not do step 1, and try to do the arecord/aplay directly I get: $ arecord -f cd | aplay -f cd imx-sgtl5000 sound.13: set sample size in capture stream first fsl-ssi-dai 2028000.ssi: ASoC: can't open interface 2028000.ssi: -11 ALSA lib pcm_dmix.c:1018:(snd_pcm_dmix_ open) unable to open slave aplay: main:660: audio open error: Device or resource busy Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo I will try to debug this, but if anyone has any suggestions, please let me know. Thanks, Fabio Estevam

2 5

[alsa-devel] Buffer size for ALSA USB PCM audio
by Alan Stern 23 Sep '13

23 Sep '13

I have been studying the data_ep_set_params() function in sound/usb/endpoint.c. This is the routine that calculates the number of samples and I/O requests to keep on the USB hardware queue for PCM audio, based on the ALSA parameters. It uses the PERIOD_BYTES parameter but not BUFFER_BYTES. In simplified terms (ignoring rounding, boundary cases, and other things), the number of periods per buffer is fixed at 24 for recording and 1 for playback, completely ignoring the user's setting. If you look at the parameters copied in snd_usb_hw_params() in sound/usb/pcm.c, you'll see what I mean. Is this really the intended behavior? It doesn't seem right at all. Alan Stern

7 44

[alsa-devel] [PATCH] ASoC: bells: Specify REFCLK for FLLs
by Charles Keepax 22 Sep '13

22 Sep '13

Wolfson recommends using a higher frequency REFCLK for best performance rather than the default 32kHz reference clock source. We should set a good example and do so. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/bells.c | 18 ++++++++++++++++++ 1 files changed, 18 insertions(+), 0 deletions(-) diff --git a/sound/soc/samsung/bells.c b/sound/soc/samsung/bells.c index 84f5d8b..89ad1d7 100644 --- a/sound/soc/samsung/bells.c +++ b/sound/soc/samsung/bells.c @@ -181,6 +181,24 @@ static int bells_late_probe(struct snd_soc_card *card) return ret; } + ret = snd_soc_codec_set_pll(codec, WM5102_FLL1_REFCLK, + ARIZONA_FLL_SRC_MCLK1, + MCLK_RATE, + bells->sysclk_rate); + if (ret != 0) { + dev_err(codec->dev, "Failed to set REFCLK for FLL1: %d\n", ret); + return ret; + } + + ret = snd_soc_codec_set_pll(codec, WM5102_FLL2_REFCLK, + ARIZONA_FLL_SRC_MCLK1, + MCLK_RATE, + bells->asyncclk_rate); + if (ret != 0) { + dev_err(codec->dev, "Failed to set REFCLK for FLL2: %d\n", ret); + return ret; + } + aif2_dai = card->rtd[DAI_CODEC_CP].cpu_dai; ret = snd_soc_dai_set_sysclk(aif2_dai, ARIZONA_CLK_ASYNCCLK, 0, 0); -- 1.7.2.5

2 3

[alsa-devel] [PATCH v2 1/2] ASoC: kirkwood: clk: probe defer when clock not yet ready
by Jean-Francois Moine 21 Sep '13

21 Sep '13

At probe time, a clock device may not be ready when some other device wants to use it. This patch lets the functions clk_get/devm_clk_get return a probe defer when the clock is defined in the DT but not yet available. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- drivers/clk/clk.c | 2 +- drivers/clk/clkdev.c | 9 +++++++-- 2 files changed, 8 insertions(+), 3 deletions(-) diff --git a/drivers/clk/clk.c b/drivers/clk/clk.c index a004769..582abc8 100644 --- a/drivers/clk/clk.c +++ b/drivers/clk/clk.c @@ -2181,7 +2181,7 @@ EXPORT_SYMBOL_GPL(of_clk_del_provider); struct clk *of_clk_get_from_provider(struct of_phandle_args *clkspec) { struct of_clk_provider *provider; - struct clk *clk = ERR_PTR(-ENOENT); + struct clk *clk = ERR_PTR(-EPROBE_DEFER); /* Check if we have such a provider in our array */ mutex_lock(&of_clk_lock); diff --git a/drivers/clk/clkdev.c b/drivers/clk/clkdev.c index 442a313..f8b28d9 100644 --- a/drivers/clk/clkdev.c +++ b/drivers/clk/clkdev.c @@ -157,8 +157,13 @@ struct clk *clk_get(struct device *dev, const char *con_id) if (dev) { clk = of_clk_get_by_name(dev->of_node, con_id); - if (!IS_ERR(clk) && __clk_get(clk)) - return clk; + if (!IS_ERR(clk)) { + if (__clk_get(clk)) + return clk; + } else { + if (PTR_ERR(clk) == -EPROBE_DEFER) + return clk; + } } return clk_get_sys(dev_id, con_id); -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 4

[alsa-devel] [PATCH v2 2/2] ASoC: kirkwood: fix loss of external clock at probe time
by Jean-Francois Moine 21 Sep '13

21 Sep '13

At probe time, when the clock driver is not yet initialized, the external clock of the kirkwood sound device will not be usable. This patch fixes this problem defering the device probe. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/kirkwood/kirkwood-i2s.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 0f3d73d..14f32d9 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -496,7 +496,10 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) return err; priv->extclk = devm_clk_get(&pdev->dev, "extclk"); - if (!IS_ERR(priv->extclk)) { + if (IS_ERR(priv->extclk)) { + if (PTR_ERR(priv->extclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + } else { if (priv->extclk == priv->clk) { devm_clk_put(&pdev->dev, priv->extclk); priv->extclk = ERR_PTR(-EINVAL); -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 1

[alsa-devel] [PATCH] ASoC: adav80x: Convert to direct regmap API usage
by Mark Brown 21 Sep '13

21 Sep '13

From: Mark Brown <broonie(a)linaro.org> As part of a move towards eliminating the ASoC level I/O code convert to use regmap directly. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/adav80x.c | 244 ++++++++++++++++++++++++++++++++++----------- 1 file changed, 186 insertions(+), 58 deletions(-) diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index 15b012d0..5d9e913 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -115,22 +115,146 @@ #define ADAV80X_PLL_OUTE_SYSCLKPD(x) BIT(2 - (x)) -static u8 adav80x_default_regs[] = { - 0x00, 0x00, 0x00, 0x00, 0x01, 0x01, 0x02, 0x01, 0x80, 0x26, 0x00, 0x00, - 0x02, 0x40, 0x20, 0x00, 0x09, 0x08, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x04, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0xd1, 0x92, 0xb1, 0x37, - 0x48, 0xd2, 0xfb, 0xca, 0xd2, 0x15, 0xe8, 0x29, 0xb9, 0x6a, 0xda, 0x2b, - 0xb7, 0xc0, 0x11, 0x65, 0x5c, 0xf6, 0xff, 0x8d, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0xa5, 0x00, 0x00, - 0x00, 0xe8, 0x46, 0xe1, 0x5b, 0xd3, 0x43, 0x77, 0x93, 0xa7, 0x44, 0xee, - 0x32, 0x12, 0xc0, 0x11, 0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0x3f, 0x3f, - 0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0x00, 0x1d, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x52, 0x00, +static const struct reg_default adav80x_default_regs[] = { + { 0, 0x00 }, + { 1, 0x00 }, + { 2, 0x00 }, + { 3, 0x00 }, + { 4, 0x01 }, + { 5, 0x01 }, + { 6, 0x02 }, + { 7, 0x01 }, + { 8, 0x80 }, + { 9, 0x26 }, + { 10, 0x00 }, + { 11, 0x00 }, + { 12, 0x02 }, + { 13, 0x40 }, + { 14, 0x20 }, + { 15, 0x00 }, + { 16, 0x09 }, + { 17, 0x08 }, + { 18, 0x00 }, + { 19, 0x00 }, + { 20, 0x00 }, + { 21, 0x00 }, + { 22, 0x00 }, + { 23, 0x00 }, + { 24, 0x04 }, + { 25, 0x00 }, + { 26, 0x01 }, + { 27, 0x00 }, + { 28, 0x00 }, + { 29, 0x00 }, + { 30, 0x00 }, + { 31, 0x00 }, + { 32, 0xd1 }, + { 33, 0x92 }, + { 34, 0xb1 }, + { 35, 0x37 }, + { 36, 0x48 }, + { 37, 0xd2 }, + { 38, 0xfb }, + { 39, 0xca }, + { 40, 0xd2 }, + { 41, 0x15 }, + { 42, 0xe8 }, + { 43, 0x29 }, + { 44, 0xb9 }, + { 45, 0x6a }, + { 46, 0xda }, + { 47, 0x2b }, + { 48, 0xb7 }, + { 49, 0xc0 }, + { 50, 0x11 }, + { 51, 0x65 }, + { 52, 0x5c }, + { 53, 0xf6 }, + { 54, 0xff }, + { 55, 0x8d }, + { 56, 0x00 }, + { 57, 0x00 }, + { 58, 0x00 }, + { 59, 0x00 }, + { 60, 0x00 }, + { 61, 0x00 }, + { 62, 0x00 }, + { 63, 0x00 }, + { 64, 0x00 }, + { 65, 0x00 }, + { 66, 0x00 }, + { 67, 0x00 }, + { 68, 0x00 }, + { 69, 0x00 }, + { 70, 0x00 }, + { 71, 0x00 }, + { 72, 0x00 }, + { 73, 0x00 }, + { 74, 0x00 }, + { 75, 0x00 }, + { 76, 0x00 }, + { 77, 0x00 }, + { 78, 0x00 }, + { 79, 0x00 }, + { 80, 0x00 }, + { 81, 0xa5 }, + { 82, 0x00 }, + { 83, 0x00 }, + { 84, 0x00 }, + { 85, 0xe8 }, + { 86, 0x46 }, + { 87, 0xe1 }, + { 88, 0x5b }, + { 89, 0xd3 }, + { 90, 0x43 }, + { 91, 0x77 }, + { 92, 0x93 }, + { 93, 0xa7 }, + { 94, 0x44 }, + { 95, 0xee }, + { 96, 0x32 }, + { 97, 0x12 }, + { 98, 0xc0 }, + { 99, 0x11 }, + { 100, 0x00 }, + { 101, 0x00 }, + { 102, 0x00 }, + { 103, 0x00 }, + { 104, 0xff }, + { 105, 0xff }, + { 106, 0x3f }, + { 107, 0x3f }, + { 108, 0x00 }, + { 109, 0x00 }, + { 110, 0x00 }, + { 111, 0x00 }, + { 112, 0xff }, + { 113, 0xff }, + { 114, 0x00 }, + { 115, 0x1d }, + { 116, 0x00 }, + { 117, 0x00 }, + { 118, 0x00 }, + { 119, 0x00 }, + { 120, 0x00 }, + { 121, 0x00 }, + { 122, 0x00 }, + { 123, 0x00 }, + { 124, 0x52 }, + { 125, 0x00 }, +}; + +static const struct regmap_config adav80x_regmap = { + .reg_bits = 7, + .val_bits = 9, + + .reg_defaults = adav80x_default_regs, + .num_reg_defaults = ARRAY_SIZE(adav80x_default_regs), + .max_register = ARRAY_SIZE(adav80x_default_regs) - 1, }; struct adav80x { - enum snd_soc_control_type control_type; + struct regmap *regmap; enum adav80x_clk_src clk_src; unsigned int sysclk; @@ -777,14 +901,9 @@ static struct snd_soc_dai_driver adav80x_dais[] = { static int adav80x_probe(struct snd_soc_codec *codec) { - int ret; struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); - ret = snd_soc_codec_set_cache_io(codec, 7, 9, adav80x->control_type); - if (ret) { - dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); - return ret; - } + codec->control_data = adav80x->regmap; /* Force PLLs on for SYSCLK output */ snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL1"); @@ -800,14 +919,25 @@ static int adav80x_probe(struct snd_soc_codec *codec) static int adav80x_suspend(struct snd_soc_codec *codec) { - return adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF); + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF); + + regcache_cache_only(adav80x->regmap, true); + + return ret; } static int adav80x_resume(struct snd_soc_codec *codec) { + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(adav80x->regmap, false); + adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - codec->cache_sync = 1; - snd_soc_cache_sync(codec); + + regcache_sync(adav80x->regmap); return 0; } @@ -827,10 +957,6 @@ static struct snd_soc_codec_driver adav80x_codec_driver = { .set_pll = adav80x_set_pll, .set_sysclk = adav80x_set_sysclk, - .reg_word_size = sizeof(u8), - .reg_cache_size = ARRAY_SIZE(adav80x_default_regs), - .reg_cache_default = adav80x_default_regs, - .controls = adav80x_controls, .num_controls = ARRAY_SIZE(adav80x_controls), .dapm_widgets = adav80x_dapm_widgets, @@ -839,34 +965,6 @@ static struct snd_soc_codec_driver adav80x_codec_driver = { .num_dapm_routes = ARRAY_SIZE(adav80x_dapm_routes), }; -static int adav80x_bus_probe(struct device *dev, - enum snd_soc_control_type control_type) -{ - struct adav80x *adav80x; - int ret; - - adav80x = kzalloc(sizeof(*adav80x), GFP_KERNEL); - if (!adav80x) - return -ENOMEM; - - dev_set_drvdata(dev, adav80x); - adav80x->control_type = control_type; - - ret = snd_soc_register_codec(dev, &adav80x_codec_driver, - adav80x_dais, ARRAY_SIZE(adav80x_dais)); - if (ret) - kfree(adav80x); - - return ret; -} - -static int adav80x_bus_remove(struct device *dev) -{ - snd_soc_unregister_codec(dev); - kfree(dev_get_drvdata(dev)); - return 0; -} - #if defined(CONFIG_SPI_MASTER) static const struct spi_device_id adav80x_spi_id[] = { { "adav801", 0 }, @@ -876,12 +974,27 @@ MODULE_DEVICE_TABLE(spi, adav80x_spi_id); static int adav80x_spi_probe(struct spi_device *spi) { - return adav80x_bus_probe(&spi->dev, SND_SOC_SPI); + struct adav80x *adav80x; + + adav80x = devm_kzalloc(&spi->dev, sizeof(*adav80x), GFP_KERNEL); + if (!adav80x) + return -ENOMEM; + + adav80x->regmap = devm_regmap_init_spi(spi, &adav80x_regmap); + if (IS_ERR(adav80x->regmap)) + return PTR_ERR(adav80x->regmap); + + dev_set_drvdata(&spi->dev, adav80x); + + return snd_soc_register_codec(&spi->dev, &adav80x_codec_driver, + adav80x_dais, ARRAY_SIZE(adav80x_dais)); } static int adav80x_spi_remove(struct spi_device *spi) { - return adav80x_bus_remove(&spi->dev); + snd_soc_unregister_codec(&spi->dev); + + return 0; } static struct spi_driver adav80x_spi_driver = { @@ -905,12 +1018,27 @@ MODULE_DEVICE_TABLE(i2c, adav80x_i2c_id); static int adav80x_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { - return adav80x_bus_probe(&client->dev, SND_SOC_I2C); + struct adav80x *adav80x; + + adav80x = devm_kzalloc(&client->dev, sizeof(*adav80x), GFP_KERNEL); + if (!adav80x) + return -ENOMEM; + + adav80x->regmap = devm_regmap_init_i2c(client, &adav80x_regmap); + if (IS_ERR(adav80x->regmap)) + return PTR_ERR(adav80x->regmap); + + dev_set_drvdata(&client->dev, adav80x); + + return snd_soc_register_codec(&client->dev, &adav80x_codec_driver, + adav80x_dais, ARRAY_SIZE(adav80x_dais)); } static int adav80x_i2c_remove(struct i2c_client *client) { - return adav80x_bus_remove(&client->dev); + snd_soc_unregister_codec(&client->dev); + + return 0; } static struct i2c_driver adav80x_i2c_driver = { -- 1.8.4.rc3

2 4

[alsa-devel] [PATCH v2 0/2] ASoC: kirkwood: fix loss of external clock at probe time
by Jean-Francois Moine 21 Sep '13

21 Sep '13

This patch series fixes the loss of clock when the requested clock device is not yet initialized at probe time. The first patch adds a -EPROBE_DEFER when the clock referenced by a phandle is not available. The second patch uses this feature to fix such a problem in the kirkwood audio driver. v2: - return the probe defer from the clock subsystem (Marc Brown remark) -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

1 0

[alsa-devel] hdmi audio on Shuttle XS35GS V3L
by Olaf Hering 20 Sep '13

20 Sep '13

Does anyone happen to know if HDMI audio is supposed to work on the Shuttle XS35GS V3L? I have tried an AOC E2260Pwhu monitor and also a Philips 22PFL3805H/12 TV, both show the picture but no sound is played via HDMI. OS is openSuSE 12.3 and 13.1beta1. Below is dmesg and /proc/asound output. Olaf ==> /proc/asound/card0/codec#0 <== Codec: IDT 92HD87B1/3 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x111d76d1 Subsystem Id: 0x12974012 Revision Id: 0x100107 No Modem Function Group found Default PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 ==> /proc/asound/card0/codec#1 <== Codec: Intel CedarTrail HDMI Address: 1 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x80862880 Subsystem Id: 0x80860101 Revision Id: 0x100000 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: ==> /proc/asound/card0/eld#1.0 <== monitor_present 0 eld_valid 0 ==> /proc/asound/card0/id <== Intel ==> /proc/asound/card0/pcm0c/info <== card: 0 device: 0 subdevice: 0 stream: CAPTURE id: 92HD87B1/3 Analog name: 92HD87B1/3 Analog subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 ==> /proc/asound/card0/pcm0c/sub0/hw_params <== closed ==> /proc/asound/card0/pcm0c/sub0/info <== card: 0 device: 0 subdevice: 0 stream: CAPTURE id: 92HD87B1/3 Analog name: 92HD87B1/3 Analog subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 ==> /proc/asound/card0/pcm0c/sub0/prealloc <== 1024 ==> /proc/asound/card0/pcm0c/sub0/prealloc_max <== 32768 ==> /proc/asound/card0/pcm0c/sub0/status <== closed ==> /proc/asound/card0/pcm0c/sub0/sw_params <== closed ==> /proc/asound/card0/pcm0c/xrun_debug <== 0 ==> /proc/asound/card0/pcm0p/info <== card: 0 device: 0 subdevice: 0 stream: PLAYBACK id: 92HD87B1/3 Analog name: 92HD87B1/3 Analog subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 ==> /proc/asound/card0/pcm0p/sub0/hw_params <== closed ==> /proc/asound/card0/pcm0p/sub0/info <== card: 0 device: 0 subdevice: 0 stream: PLAYBACK id: 92HD87B1/3 Analog name: 92HD87B1/3 Analog subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 ==> /proc/asound/card0/pcm0p/sub0/prealloc <== 1024 ==> /proc/asound/card0/pcm0p/sub0/prealloc_max <== 32768 ==> /proc/asound/card0/pcm0p/sub0/status <== closed ==> /proc/asound/card0/pcm0p/sub0/sw_params <== closed ==> /proc/asound/card0/pcm0p/xrun_debug <== 0 ==> /proc/asound/card0/pcm3p/info <== card: 0 device: 3 subdevice: 0 stream: PLAYBACK id: HDMI 0 name: HDMI 0 subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 ==> /proc/asound/card0/pcm3p/sub0/hw_params <== closed ==> /proc/asound/card0/pcm3p/sub0/info <== card: 0 device: 3 subdevice: 0 stream: PLAYBACK id: HDMI 0 name: HDMI 0 subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 ==> /proc/asound/card0/pcm3p/sub0/prealloc <== 1024 ==> /proc/asound/card0/pcm3p/sub0/prealloc_max <== 32768 ==> /proc/asound/card0/pcm3p/sub0/status <== closed ==> /proc/asound/card0/pcm3p/sub0/sw_params <== closed ==> /proc/asound/card0/pcm3p/xrun_debug <== 0 ==> /proc/asound/card1/codec#0 <== Codec: ATI R6xx HDMI Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x1002aa01 Subsystem Id: 0x00aa0100 Revision Id: 0x100200 No Modem Function Group found Default PCM: rates [0x70]: 32000 44100 48000 bits [0x2]: 16 ==> /proc/asound/card1/eld#0.0 <== monitor_present 0 eld_valid 0 ==> /proc/asound/card1/id <== Generic ==> /proc/asound/card1/pcm3p/info <== card: 1 device: 3 subdevice: 0 stream: PLAYBACK id: HDMI 0 name: HDMI 0 subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 ==> /proc/asound/card1/pcm3p/sub0/hw_params <== closed ==> /proc/asound/card1/pcm3p/sub0/info <== card: 1 device: 3 subdevice: 0 stream: PLAYBACK id: HDMI 0 name: HDMI 0 subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 ==> /proc/asound/card1/pcm3p/sub0/prealloc <== 1024 ==> /proc/asound/card1/pcm3p/sub0/prealloc_max <== 32768 ==> /proc/asound/card1/pcm3p/sub0/status <== closed ==> /proc/asound/card1/pcm3p/sub0/sw_params <== closed ==> /proc/asound/card1/pcm3p/xrun_debug <== 0 ==> /proc/asound/cards <== 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xd0300000 irq 47 1 [Generic ]: HDA-Intel - HD-Audio Generic HD-Audio Generic at 0xd0040000 irq 48 ==> /proc/asound/devices <== 1: : sequencer 2: [ 0- 3]: digital audio playback 3: [ 0- 0]: digital audio playback 4: [ 0- 0]: digital audio capture 5: [ 0- 1]: hardware dependent 6: [ 0- 0]: hardware dependent 7: [ 0] : control 8: [ 1- 3]: digital audio playback 9: [ 1- 0]: hardware dependent 10: [ 1] : control ==> /proc/asound/hwdep <== 00-01: HDA Codec 1 00-00: HDA Codec 0 01-00: HDA Codec 0 ==> /proc/asound/modules <== 0 snd_hda_intel 1 snd_hda_intel ==> /proc/asound/oss/devices <== ==> /proc/asound/oss/sndstat <== Sound Driver:3.8.1a-980706 (ALSA emulation code) Kernel: Linux shuttle 3.11.1-1.g1383321-desktop #1 SMP PREEMPT Sat Sep 14 18:49:04 UTC 2013 (1383321) x86_64 Config options: 0 Installed drivers: Type 10: ALSA emulation Card config: HDA Intel at 0xd0300000 irq 47 HD-Audio Generic at 0xd0040000 irq 48 ==> /proc/asound/pcm <== 00-00: 92HD87B1/3 Analog : 92HD87B1/3 Analog : playback 1 : capture 1 00-03: HDMI 0 : HDMI 0 : playback 1 01-03: HDMI 0 : HDMI 0 : playback 1 ==> /proc/asound/seq/clients <== Client info cur clients : 2 peak clients : 2 max clients : 192 Client 0 : "System" [Kernel] Port 0 : "Timer" (Rwe-) Port 1 : "Announce" (R-e-) Client 14 : "Midi Through" [Kernel] Port 0 : "Midi Through Port-0" (RWe-) ==> /proc/asound/seq/drivers <== ==> /proc/asound/seq/queues <== ==> /proc/asound/seq/timer <== ==> /proc/asound/timers <== G0: system timer : 1000.000us (10000000 ticks) P0-0-0: PCM playback 0-0-0 : SLAVE P0-0-1: PCM capture 0-0-1 : SLAVE P0-3-0: PCM playback 0-3-0 : SLAVE P1-3-0: PCM playback 1-3-0 : SLAVE ==> /proc/asound/version <== Advanced Linux Sound Architecture Driver Version k3.11.1-1.g1383321-desktop. shuttle:~ # dmesg [ 0.000000] Initializing cgroup subsys cpuset [ 0.000000] Initializing cgroup subsys cpu [ 0.000000] Initializing cgroup subsys cpuacct [ 0.000000] Linux version 3.11.1-1.g1383321-desktop (geeko@buildhost) (gcc version 4.8.1 20130909 [gcc-4_8-branch revision 202388] (SUSE Linux) ) #1 SMP PREEMPT Sat Sep 14 18:49:04 UTC 2013 (1383321) [ 0.000000] Command line: BOOT_IMAGE=/boot/vmlinuz-3.11.1-1.g1383321-desktop root=UUID=b8a2a174-296a-4da2-82e3-9680fadcff42 quiet panic=9 sysrq=yes dhcp=1 resume=/dev/disk/by-id/ata-KINGSTON_SV300S37A60G_50026B723106E9DA-part1 splash=silent quiet showopts [ 0.000000] Disabled fast string operations [ 0.000000] e820: BIOS-provided physical RAM map: [ 0.000000] BIOS-e820: [mem 0x0000000000000000-0x000000000009fbff] usable [ 0.000000] BIOS-e820: [mem 0x000000000009fc00-0x000000000009ffff] reserved [ 0.000000] BIOS-e820: [mem 0x00000000000e0000-0x00000000000fffff] reserved [ 0.000000] BIOS-e820: [mem 0x0000000000100000-0x00000000bf7f3fff] usable [ 0.000000] BIOS-e820: [mem 0x00000000bf7f4000-0x00000000bf877fff] ACPI NVS [ 0.000000] BIOS-e820: [mem 0x00000000bf878000-0x00000000bf888fff] reserved [ 0.000000] BIOS-e820: [mem 0x00000000bf889000-0x00000000bf88ffff] ACPI data [ 0.000000] BIOS-e820: [mem 0x00000000bf890000-0x00000000bf8abfff] reserved [ 0.000000] BIOS-e820: [mem 0x00000000bf8ac000-0x00000000bf8acfff] usable [ 0.000000] BIOS-e820: [mem 0x00000000bf8ad000-0x00000000bf8cafff] reserved [ 0.000000] BIOS-e820: [mem 0x00000000bf8cb000-0x00000000bf8ccfff] usable [ 0.000000] BIOS-e820: [mem 0x00000000bf8cd000-0x00000000bf8dbfff] ACPI NVS [ 0.000000] BIOS-e820: [mem 0x00000000bf8dc000-0x00000000bf901fff] reserved [ 0.000000] BIOS-e820: [mem 0x00000000bf902000-0x00000000bf944fff] ACPI NVS [ 0.000000] BIOS-e820: [mem 0x00000000bf945000-0x00000000bfd1afff] usable [ 0.000000] BIOS-e820: [mem 0x00000000bfd1b000-0x00000000bfee4fff] reserved [ 0.000000] BIOS-e820: [mem 0x00000000bfee5000-0x00000000bfeeffff] usable [ 0.000000] BIOS-e820: [mem 0x00000000bfef0000-0x00000000bfffffff] reserved [ 0.000000] BIOS-e820: [mem 0x00000000e0000000-0x00000000efffffff] reserved [ 0.000000] BIOS-e820: [mem 0x00000000fec00000-0x00000000fec00fff] reserved [ 0.000000] BIOS-e820: [mem 0x00000000fed00000-0x00000000fed00fff] reserved [ 0.000000] BIOS-e820: [mem 0x00000000fed1c000-0x00000000fed8ffff] reserved [ 0.000000] BIOS-e820: [mem 0x00000000fee00000-0x00000000fee00fff] reserved [ 0.000000] BIOS-e820: [mem 0x00000000ffe00000-0x00000000ffffffff] reserved [ 0.000000] BIOS-e820: [mem 0x0000000100000000-0x000000013fffffff] usable [ 0.000000] NX (Execute Disable) protection: active [ 0.000000] SMBIOS 2.7 present. [ 0.000000] DMI: Shuttle Inc. XS35V3/XS35V3, BIOS 1.11 12/13/2012 [ 0.000000] e820: update [mem 0x00000000-0x00000fff] usable ==> reserved [ 0.000000] e820: remove [mem 0x000a0000-0x000fffff] usable [ 0.000000] No AGP bridge found [ 0.000000] e820: last_pfn = 0x140000 max_arch_pfn = 0x400000000 [ 0.000000] MTRR default type: uncachable [ 0.000000] MTRR fixed ranges enabled: [ 0.000000] 00000-9FFFF write-back [ 0.000000] A0000-BFFFF uncachable [ 0.000000] C0000-E7FFF write-through [ 0.000000] E8000-FFFFF write-protect [ 0.000000] MTRR variable ranges enabled: [ 0.000000] 0 base 000000000 mask F00000000 write-back [ 0.000000] 1 base 100000000 mask FC0000000 write-back [ 0.000000] 2 base 0BFF00000 mask FFFF00000 write-through [ 0.000000] 3 base 0C0000000 mask FC0000000 uncachable [ 0.000000] 4 disabled [ 0.000000] 5 disabled [ 0.000000] 6 disabled [ 0.000000] x86 PAT enabled: cpu 0, old 0x7040600070406, new 0x7010600070106 [ 0.000000] e820: last_pfn = 0xbfef0 max_arch_pfn = 0x400000000 [ 0.000000] found SMP MP-table at [mem 0x000fc9c0-0x000fc9cf] mapped at [ffff8800000fc9c0] [ 0.000000] Scanning 1 areas for low memory corruption [ 0.000000] Base memory trampoline at [ffff880000099000] 99000 size 24576 [ 0.000000] init_memory_mapping: [mem 0x00000000-0x000fffff] [ 0.000000] [mem 0x00000000-0x000fffff] page 4k [ 0.000000] BRK [0x01fb3000, 0x01fb3fff] PGTABLE [ 0.000000] BRK [0x01fb4000, 0x01fb4fff] PGTABLE [ 0.000000] BRK [0x01fb5000, 0x01fb5fff] PGTABLE [ 0.000000] init_memory_mapping: [mem 0x13fe00000-0x13fffffff] [ 0.000000] [mem 0x13fe00000-0x13fffffff] page 2M [ 0.000000] BRK [0x01fb6000, 0x01fb6fff] PGTABLE [ 0.000000] init_memory_mapping: [mem 0x13c000000-0x13fdfffff] [ 0.000000] [mem 0x13c000000-0x13fdfffff] page 2M [ 0.000000] init_memory_mapping: [mem 0x100000000-0x13bffffff] [ 0.000000] [mem 0x100000000-0x13bffffff] page 2M [ 0.000000] init_memory_mapping: [mem 0x00100000-0xbf7f3fff] [ 0.000000] [mem 0x00100000-0x001fffff] page 4k [ 0.000000] [mem 0x00200000-0xbf5fffff] page 2M [ 0.000000] [mem 0xbf600000-0xbf7f3fff] page 4k [ 0.000000] init_memory_mapping: [mem 0xbf8ac000-0xbf8acfff] [ 0.000000] [mem 0xbf8ac000-0xbf8acfff] page 4k [ 0.000000] BRK [0x01fb7000, 0x01fb7fff] PGTABLE [ 0.000000] init_memory_mapping: [mem 0xbf8cb000-0xbf8ccfff] [ 0.000000] [mem 0xbf8cb000-0xbf8ccfff] page 4k [ 0.000000] init_memory_mapping: [mem 0xbf945000-0xbfd1afff] [ 0.000000] [mem 0xbf945000-0xbf9fffff] page 4k [ 0.000000] [mem 0xbfa00000-0xbfbfffff] page 2M [ 0.000000] [mem 0xbfc00000-0xbfd1afff] page 4k [ 0.000000] BRK [0x01fb8000, 0x01fb8fff] PGTABLE [ 0.000000] init_memory_mapping: [mem 0xbfee5000-0xbfeeffff] [ 0.000000] [mem 0xbfee5000-0xbfeeffff] page 4k [ 0.000000] RAMDISK: [mem 0x34eee000-0x3676efff] [ 0.000000] ACPI: RSDP 00000000000f0450 00024 (v02 Shuttl) [ 0.000000] ACPI: XSDT 00000000bf889078 0006C (v01 Shuttl Shuttle 01072009 AMI 00010013) [ 0.000000] ACPI: FACP 00000000bf88ea08 000F4 (v04 Shuttl Shuttle 01072009 AMI 00010013) [ 0.000000] ACPI: DSDT 00000000bf889170 05894 (v02 Shuttl XS35V300 00000000 INTL 20051117) [ 0.000000] ACPI: FACS 00000000bf8d9f80 00040 [ 0.000000] ACPI: APIC 00000000bf88eb00 00072 (v03 Shuttl Shuttle 01072009 AMI 00010013) [ 0.000000] ACPI: MCFG 00000000bf88eb78 0003C (v01 Shuttl Shuttle 01072009 MSFT 00000097) [ 0.000000] ACPI: SLIC 00000000bf88ebb8 00176 (v01 Shuttl Shuttle 01072009 AMI 00010013) [ 0.000000] ACPI: HPET 00000000bf88ed30 00038 (v01 Shuttl Shuttle 01072009 AMI. 00000005) [ 0.000000] ACPI: SSDT 00000000bf88ed68 00655 (v01 PmRef CpuPm 00003000 INTL 20051117) [ 0.000000] ACPI: SSDT 00000000bf88f3c0 00259 (v01 PmRef Cpu0Tst 00003000 INTL 20051117) [ 0.000000] ACPI: SSDT 00000000bf88f620 0020F (v01 PmRef ApTst 00003000 INTL 20051117) [ 0.000000] ACPI: IFEU 00000000bf88f830 00042 (v01 Shuttl Shuttle 01072009 00000000) [ 0.000000] ACPI: Local APIC address 0xfee00000 [ 0.000000] No NUMA configuration found [ 0.000000] Faking a node at [mem 0x0000000000000000-0x000000013fffffff] [ 0.000000] Initmem setup node 0 [mem 0x00000000-0x13fffffff] [ 0.000000] NODE_DATA [mem 0x13ffe8000-0x13fffbfff] [ 0.000000] [ffffea0000000000-ffffea00045fffff] PMD -> [ffff88013b600000-ffff88013edfffff] on node 0 [ 0.000000] Zone ranges: [ 0.000000] DMA [mem 0x00001000-0x00ffffff] [ 0.000000] DMA32 [mem 0x01000000-0xffffffff] [ 0.000000] Normal [mem 0x100000000-0x13fffffff] [ 0.000000] Movable zone start for each node [ 0.000000] Early memory node ranges [ 0.000000] node 0: [mem 0x00001000-0x0009efff] [ 0.000000] node 0: [mem 0x00100000-0xbf7f3fff] [ 0.000000] node 0: [mem 0xbf8ac000-0xbf8acfff] [ 0.000000] node 0: [mem 0xbf8cb000-0xbf8ccfff] [ 0.000000] node 0: [mem 0xbf945000-0xbfd1afff] [ 0.000000] node 0: [mem 0xbfee5000-0xbfeeffff] [ 0.000000] node 0: [mem 0x100000000-0x13fffffff] [ 0.000000] On node 0 totalpages: 1047414 [ 0.000000] DMA zone: 56 pages used for memmap [ 0.000000] DMA zone: 21 pages reserved [ 0.000000] DMA zone: 3998 pages, LIFO batch:0 [ 0.000000] DMA32 zone: 10682 pages used for memmap [ 0.000000] DMA32 zone: 781272 pages, LIFO batch:31 [ 0.000000] Normal zone: 3584 pages used for memmap [ 0.000000] Normal zone: 262144 pages, LIFO batch:31 [ 0.000000] ACPI: PM-Timer IO Port: 0x408 [ 0.000000] ACPI: Local APIC address 0xfee00000 [ 0.000000] ACPI: LAPIC (acpi_id[0x01] lapic_id[0x00] enabled) [ 0.000000] ACPI: LAPIC (acpi_id[0x02] lapic_id[0x01] enabled) [ 0.000000] ACPI: LAPIC (acpi_id[0x03] lapic_id[0x02] enabled) [ 0.000000] ACPI: LAPIC (acpi_id[0x04] lapic_id[0x03] enabled) [ 0.000000] ACPI: LAPIC_NMI (acpi_id[0xff] high edge lint[0x1]) [ 0.000000] ACPI: IOAPIC (id[0x04] address[0xfec00000] gsi_base[0]) [ 0.000000] IOAPIC[0]: apic_id 4, version 32, address 0xfec00000, GSI 0-23 [ 0.000000] ACPI: INT_SRC_OVR (bus 0 bus_irq 0 global_irq 2 dfl dfl) [ 0.000000] ACPI: INT_SRC_OVR (bus 0 bus_irq 9 global_irq 9 high level) [ 0.000000] ACPI: IRQ0 used by override. [ 0.000000] ACPI: IRQ2 used by override. [ 0.000000] ACPI: IRQ9 used by override. [ 0.000000] Using ACPI (MADT) for SMP configuration information [ 0.000000] ACPI: HPET id: 0x8086a201 base: 0xfed00000 [ 0.000000] smpboot: Allowing 4 CPUs, 0 hotplug CPUs [ 0.000000] nr_irqs_gsi: 40 [ 0.000000] PM: Registered nosave memory: [mem 0x0009f000-0x0009ffff] [ 0.000000] PM: Registered nosave memory: [mem 0x000a0000-0x000dffff] [ 0.000000] PM: Registered nosave memory: [mem 0x000e0000-0x000fffff] [ 0.000000] PM: Registered nosave memory: [mem 0xbf7f4000-0xbf877fff] [ 0.000000] PM: Registered nosave memory: [mem 0xbf878000-0xbf888fff] [ 0.000000] PM: Registered nosave memory: [mem 0xbf889000-0xbf88ffff] [ 0.000000] PM: Registered nosave memory: [mem 0xbf890000-0xbf8abfff] [ 0.000000] PM: Registered nosave memory: [mem 0xbf8ad000-0xbf8cafff] [ 0.000000] PM: Registered nosave memory: [mem 0xbf8cd000-0xbf8dbfff] [ 0.000000] PM: Registered nosave memory: [mem 0xbf8dc000-0xbf901fff] [ 0.000000] PM: Registered nosave memory: [mem 0xbf902000-0xbf944fff] [ 0.000000] PM: Registered nosave memory: [mem 0xbfd1b000-0xbfee4fff] [ 0.000000] PM: Registered nosave memory: [mem 0xbfef0000-0xbfffffff] [ 0.000000] PM: Registered nosave memory: [mem 0xc0000000-0xdfffffff] [ 0.000000] PM: Registered nosave memory: [mem 0xe0000000-0xefffffff] [ 0.000000] PM: Registered nosave memory: [mem 0xf0000000-0xfebfffff] [ 0.000000] PM: Registered nosave memory: [mem 0xfec00000-0xfec00fff] [ 0.000000] PM: Registered nosave memory: [mem 0xfec01000-0xfecfffff] [ 0.000000] PM: Registered nosave memory: [mem 0xfed00000-0xfed00fff] [ 0.000000] PM: Registered nosave memory: [mem 0xfed01000-0xfed1bfff] [ 0.000000] PM: Registered nosave memory: [mem 0xfed1c000-0xfed8ffff] [ 0.000000] PM: Registered nosave memory: [mem 0xfed90000-0xfedfffff] [ 0.000000] PM: Registered nosave memory: [mem 0xfee00000-0xfee00fff] [ 0.000000] PM: Registered nosave memory: [mem 0xfee01000-0xffdfffff] [ 0.000000] PM: Registered nosave memory: [mem 0xffe00000-0xffffffff] [ 0.000000] e820: [mem 0xc0000000-0xdfffffff] available for PCI devices [ 0.000000] Booting paravirtualized kernel on bare hardware [ 0.000000] setup_percpu: NR_CPUS:512 nr_cpumask_bits:512 nr_cpu_ids:4 nr_node_ids:1 [ 0.000000] PERCPU: Embedded 28 pages/cpu @ffff88013fc00000 s83584 r8192 d22912 u524288 [ 0.000000] pcpu-alloc: s83584 r8192 d22912 u524288 alloc=1*2097152 [ 0.000000] pcpu-alloc: [0] 0 1 2 3 [ 0.000000] Built 1 zonelists in Node order, mobility grouping on. Total pages: 1033071 [ 0.000000] Policy zone: Normal [ 0.000000] Kernel command line: BOOT_IMAGE=/boot/vmlinuz-3.11.1-1.g1383321-desktop root=UUID=b8a2a174-296a-4da2-82e3-9680fadcff42 quiet panic=9 sysrq=yes dhcp=1 resume=/dev/disk/by-id/ata-KINGSTON_SV300S37A60G_50026B723106E9DA-part1 splash=silent quiet showopts [ 0.000000] PID hash table entries: 4096 (order: 3, 32768 bytes) [ 0.000000] Checking aperture... [ 0.000000] No AGP bridge found [ 0.000000] Memory: 4024188K/4189656K available (5868K kernel code, 778K rwdata, 4368K rodata, 1332K init, 1640K bss, 165468K reserved) [ 0.000000] Preemptible hierarchical RCU implementation. [ 0.000000] RCU dyntick-idle grace-period acceleration is enabled. [ 0.000000] RCU restricting CPUs from NR_CPUS=512 to nr_cpu_ids=4. [ 0.000000] Offload RCU callbacks from all CPUs [ 0.000000] Offload RCU callbacks from CPUs: 0-511. [ 0.000000] NR_IRQS:33024 nr_irqs:712 16 [ 0.000000] spurious 8259A interrupt: IRQ7. [ 0.000000] Console: colour dummy device 80x25 [ 0.000000] console [tty0] enabled [ 0.000000] allocated 16777216 bytes of page_cgroup [ 0.000000] please try 'cgroup_disable=memory' option if you don't want memory cgroups [ 0.000000] hpet clockevent registered [ 0.000000] tsc: Fast TSC calibration using PIT [ 0.001000] tsc: Detected 1861.931 MHz processor [ 0.000006] Calibrating delay loop (skipped), value calculated using timer frequency.. 3723.86 BogoMIPS (lpj=1861931) [ 0.000012] pid_max: default: 32768 minimum: 301 [ 0.000105] Security Framework initialized [ 0.000129] AppArmor: AppArmor initialized [ 0.000790] Dentry cache hash table entries: 524288 (order: 10, 4194304 bytes) [ 0.004142] Inode-cache hash table entries: 262144 (order: 9, 2097152 bytes) [ 0.005662] Mount-cache hash table entries: 256 [ 0.006164] Initializing cgroup subsys memory [ 0.006197] Initializing cgroup subsys devices [ 0.006203] Initializing cgroup subsys freezer [ 0.006207] Initializing cgroup subsys net_cls [ 0.006211] Initializing cgroup subsys blkio [ 0.006215] Initializing cgroup subsys perf_event [ 0.006222] Initializing cgroup subsys hugetlb [ 0.006263] Disabled fast string operations [ 0.006271] CPU: Physical Processor ID: 0 [ 0.006274] CPU: Processor Core ID: 0 [ 0.006278] mce: CPU supports 5 MCE banks [ 0.006289] CPU0: Thermal monitoring enabled (TM1) [ 0.006301] Last level iTLB entries: 4KB 32, 2MB 0, 4MB 0 Last level dTLB entries: 4KB 64, 2MB 0, 4MB 8 tlb_flushall_shift: 6 [ 0.006426] Freeing SMP alternatives memory: 24K (ffffffff81e11000 - ffffffff81e17000) [ 0.007539] ACPI: Core revision 20130517 [ 0.020130] ACPI: All ACPI Tables successfully acquired [ 0.021223] ..TIMER: vector=0x30 apic1=0 pin1=2 apic2=0 pin2=0 [ 0.031231] smpboot: CPU0: Intel(R) Atom(TM) CPU D2550 @ 1.86GHz (fam: 06, model: 36, stepping: 01) [ 0.132916] Performance Events: PEBS fmt0+, 8-deep LBR, Atom events, Intel PMU driver. [ 0.132929] ... version: 3 [ 0.132932] ... bit width: 40 [ 0.132935] ... generic registers: 2 [ 0.132938] ... value mask: 000000ffffffffff [ 0.132941] ... max period: 000000007fffffff [ 0.132943] ... fixed-purpose events: 3 [ 0.132946] ... event mask: 0000000700000003 [ 0.155179] Disabled fast string operations [ 0.157493] NMI watchdog: enabled on all CPUs, permanently consumes one hw-PMU counter. [ 0.170159] Disabled fast string operations [ 0.144138] smpboot: Booting Node 0, Processors #1 #2 #3 OK [ 0.185160] Disabled fast string operations [ 0.187336] Brought up 4 CPUs [ 0.187345] smpboot: Total of 4 processors activated (14895.44 BogoMIPS) [ 0.188501] devtmpfs: initialized [ 0.192382] PM: Registering ACPI NVS region [mem 0xbf7f4000-0xbf877fff] (540672 bytes) [ 0.192426] PM: Registering ACPI NVS region [mem 0xbf8cd000-0xbf8dbfff] (61440 bytes) [ 0.192434] PM: Registering ACPI NVS region [mem 0xbf902000-0xbf944fff] (274432 bytes) [ 0.192811] RTC time: 14:10:10, date: 09/19/13 [ 0.192959] NET: Registered protocol family 16 [ 0.193372] ACPI: bus type PCI registered [ 0.193379] acpiphp: ACPI Hot Plug PCI Controller Driver version: 0.5 [ 0.193516] PCI: MMCONFIG for domain 0000 [bus 00-ff] at [mem 0xe0000000-0xefffffff] (base 0xe0000000) [ 0.193523] PCI: MMCONFIG at [mem 0xe0000000-0xefffffff] reserved in E820 [ 0.229808] PCI: Using configuration type 1 for base access [ 0.232752] bio: create slab <bio-0> at 0 [ 0.233147] ACPI: Added _OSI(Module Device) [ 0.233154] ACPI: Added _OSI(Processor Device) [ 0.233158] ACPI: Added _OSI(3.0 _SCP Extensions) [ 0.233163] ACPI: Added _OSI(Processor Aggregator Device) [ 0.235600] ACPI: EC: Look up EC in DSDT [ 0.238399] ACPI: Executed 1 blocks of module-level executable AML code [ 0.248370] [Firmware Bug]: ACPI: BIOS _OSI(Linux) query ignored [ 0.252986] ACPI: Interpreter enabled [ 0.253014] ACPI Exception: AE_NOT_FOUND, While evaluating Sleep State [\_S2_] (20130517/hwxface-571) [ 0.253053] ACPI: (supports S0 S1 S3 S4 S5) [ 0.253057] ACPI: Using IOAPIC for interrupt routing [ 0.253158] PCI: Using host bridge windows from ACPI; if necessary, use "pci=nocrs" and report a bug [ 0.253496] ACPI: No dock devices found. [ 0.273701] ACPI: Power Resource [FN00] (off) [ 0.275547] ACPI: PCI Root Bridge [PCI0] (domain 0000 [bus 00-ff]) [ 0.275734] \_SB_.PCI0:_OSC invalid UUID [ 0.275739] _OSC request data:1 8 0 [ 0.276882] PCI host bridge to bus 0000:00 [ 0.276891] pci_bus 0000:00: root bus resource [bus 00-ff] [ 0.276898] pci_bus 0000:00: root bus resource [io 0x0000-0x0cf7] [ 0.276903] pci_bus 0000:00: root bus resource [io 0x0d00-0xffff] [ 0.276920] pci_bus 0000:00: root bus resource [mem 0x000a0000-0x000bffff] [ 0.276925] pci_bus 0000:00: root bus resource [mem 0x000c0000-0x000dffff] [ 0.276931] pci_bus 0000:00: root bus resource [mem 0x000e0000-0x000effff] [ 0.276936] pci_bus 0000:00: root bus resource [mem 0x000f0000-0x000fffff] [ 0.276941] pci_bus 0000:00: root bus resource [mem 0xc0000000-0xfebfffff] [ 0.276958] pci 0000:00:00.0: [8086:0bf3] type 00 class 0x060000 [ 0.277240] pci 0000:00:1b.0: [8086:27d8] type 00 class 0x040300 [ 0.277274] pci 0000:00:1b.0: reg 0x10: [mem 0xd0300000-0xd0303fff 64bit] [ 0.277404] pci 0000:00:1b.0: PME# supported from D0 D3hot D3cold [ 0.277619] pci 0000:00:1c.0: [8086:27d0] type 01 class 0x060400 [ 0.277757] pci 0000:00:1c.0: PME# supported from D0 D3hot D3cold [ 0.277901] pci 0000:00:1c.0: System wakeup disabled by ACPI [ 0.278003] pci 0000:00:1c.1: [8086:27d2] type 01 class 0x060400 [ 0.278139] pci 0000:00:1c.1: PME# supported from D0 D3hot D3cold [ 0.278286] pci 0000:00:1c.1: System wakeup disabled by ACPI [ 0.278383] pci 0000:00:1c.3: [8086:27d6] type 01 class 0x060400 [ 0.278518] pci 0000:00:1c.3: PME# supported from D0 D3hot D3cold [ 0.278660] pci 0000:00:1c.3: System wakeup disabled by ACPI [ 0.278754] pci 0000:00:1d.0: [8086:27c8] type 00 class 0x0c0300 [ 0.278827] pci 0000:00:1d.0: reg 0x20: [io 0xf0a0-0xf0bf] [ 0.279052] pci 0000:00:1d.0: System wakeup disabled by ACPI [ 0.279145] pci 0000:00:1d.1: [8086:27c9] type 00 class 0x0c0300 [ 0.279216] pci 0000:00:1d.1: reg 0x20: [io 0xf080-0xf09f] [ 0.279434] pci 0000:00:1d.1: System wakeup disabled by ACPI [ 0.279522] pci 0000:00:1d.2: [8086:27ca] type 00 class 0x0c0300 [ 0.279593] pci 0000:00:1d.2: reg 0x20: [io 0xf060-0xf07f] [ 0.279801] pci 0000:00:1d.2: System wakeup disabled by ACPI [ 0.279888] pci 0000:00:1d.3: [8086:27cb] type 00 class 0x0c0300 [ 0.279969] pci 0000:00:1d.3: reg 0x20: [io 0xf040-0xf05f] [ 0.280177] pci 0000:00:1d.3: System wakeup disabled by ACPI [ 0.280279] pci 0000:00:1d.7: [8086:27cc] type 00 class 0x0c0320 [ 0.280314] pci 0000:00:1d.7: reg 0x10: [mem 0xd0305000-0xd03053ff] [ 0.280444] pci 0000:00:1d.7: PME# supported from D0 D3hot D3cold [ 0.280579] pci 0000:00:1d.7: System wakeup disabled by ACPI [ 0.280673] pci 0000:00:1e.0: [8086:2448] type 01 class 0x060401 [ 0.280893] pci 0000:00:1e.0: System wakeup disabled by ACPI [ 0.280993] pci 0000:00:1f.0: [8086:27bc] type 00 class 0x060100 [ 0.281327] pci 0000:00:1f.2: [8086:27c1] type 00 class 0x010601 [ 0.281357] pci 0000:00:1f.2: reg 0x10: [io 0xf0f0-0xf0f7] [ 0.281373] pci 0000:00:1f.2: reg 0x14: [io 0xf0e0-0xf0e3] [ 0.281391] pci 0000:00:1f.2: reg 0x18: [io 0xf0d0-0xf0d7] [ 0.281407] pci 0000:00:1f.2: reg 0x1c: [io 0xf0c0-0xf0c3] [ 0.281425] pci 0000:00:1f.2: reg 0x20: [io 0xf020-0xf02f] [ 0.281442] pci 0000:00:1f.2: reg 0x24: [mem 0xd0304000-0xd03043ff] [ 0.281516] pci 0000:00:1f.2: PME# supported from D3hot [ 0.281716] pci 0000:00:1f.3: [8086:27da] type 00 class 0x0c0500 [ 0.281788] pci 0000:00:1f.3: reg 0x20: [io 0xf000-0xf01f] [ 0.282172] pci 0000:01:00.0: [1002:6761] type 00 class 0x030000 [ 0.282208] pci 0000:01:00.0: reg 0x10: [mem 0xc0000000-0xcfffffff 64bit pref] [ 0.282235] pci 0000:01:00.0: reg 0x18: [mem 0xd0020000-0xd003ffff 64bit] [ 0.282254] pci 0000:01:00.0: reg 0x20: [io 0xe000-0xe0ff] [ 0.282286] pci 0000:01:00.0: reg 0x30: [mem 0xd0000000-0xd001ffff pref] [ 0.282367] pci 0000:01:00.0: supports D1 D2 [ 0.282425] pci 0000:01:00.0: System wakeup disabled by ACPI [ 0.282527] pci 0000:01:00.1: [1002:aa98] type 00 class 0x040300 [ 0.282562] pci 0000:01:00.1: reg 0x10: [mem 0xd0040000-0xd0043fff 64bit] [ 0.282707] pci 0000:01:00.1: supports D1 D2 [ 0.283939] pci 0000:00:1c.0: PCI bridge to [bus 01] [ 0.283949] pci 0000:00:1c.0: bridge window [io 0xe000-0xefff] [ 0.283958] pci 0000:00:1c.0: bridge window [mem 0xc0000000-0xd00fffff] [ 0.284221] acpiphp: Slot [1] registered [ 0.284298] pci 0000:02:00.0: [10ec:5289] type 00 class 0xff0000 [ 0.284326] pci 0000:02:00.0: reg 0x10: [mem 0xd0200000-0xd020ffff] [ 0.284496] pci 0000:02:00.0: supports D1 D2 [ 0.284501] pci 0000:02:00.0: PME# supported from D1 D2 D3hot D3cold [ 0.284583] pci 0000:02:00.0: System wakeup disabled by ACPI [ 0.284695] pci 0000:02:00.2: [10ec:8168] type 00 class 0x020000 [ 0.284721] pci 0000:02:00.2: reg 0x10: [io 0xd000-0xd0ff] [ 0.284761] pci 0000:02:00.2: reg 0x18: [mem 0xd0214000-0xd0214fff 64bit pref] [ 0.284788] pci 0000:02:00.2: reg 0x20: [mem 0xd0210000-0xd0213fff 64bit pref] [ 0.284891] pci 0000:02:00.2: supports D1 D2 [ 0.284896] pci 0000:02:00.2: PME# supported from D0 D1 D2 D3hot D3cold [ 0.286942] pci 0000:00:1c.1: PCI bridge to [bus 02] [ 0.286952] pci 0000:00:1c.1: bridge window [io 0xd000-0xdfff] [ 0.286960] pci 0000:00:1c.1: bridge window [mem 0xd0200000-0xd02fffff] [ 0.287110] pci 0000:00:1c.3: PCI bridge to [bus 03] [ 0.287120] pci 0000:00:1c.3: bridge window [io 0xc000-0xcfff] [ 0.287128] pci 0000:00:1c.3: bridge window [mem 0xd0100000-0xd01fffff] [ 0.287280] pci 0000:00:1e.0: PCI bridge to [bus 04] (subtractive decode) [ 0.287302] pci 0000:00:1e.0: bridge window [io 0x0000-0x0cf7] (subtractive decode) [ 0.287308] pci 0000:00:1e.0: bridge window [io 0x0d00-0xffff] (subtractive decode) [ 0.287314] pci 0000:00:1e.0: bridge window [mem 0x000a0000-0x000bffff] (subtractive decode) [ 0.287319] pci 0000:00:1e.0: bridge window [mem 0x000c0000-0x000dffff] (subtractive decode) [ 0.287325] pci 0000:00:1e.0: bridge window [mem 0x000e0000-0x000effff] (subtractive decode) [ 0.287330] pci 0000:00:1e.0: bridge window [mem 0x000f0000-0x000fffff] (subtractive decode) [ 0.287335] pci 0000:00:1e.0: bridge window [mem 0xc0000000-0xfebfffff] (subtractive decode) [ 0.287573] \_SB_.PCI0:_OSC invalid UUID [ 0.287577] _OSC request data:1 1f 0 [ 0.287587] acpi PNP0A08:00: ACPI _OSC support notification failed, disabling PCIe ASPM [ 0.287592] acpi PNP0A08:00: Unable to request _OSC control (_OSC support mask: 0x08) [ 0.288565] ACPI: PCI Interrupt Link [LNKA] (IRQs 1 3 4 5 6 7 *10 12 14 15) [ 0.288824] ACPI: PCI Interrupt Link [LNKB] (IRQs 1 3 4 *5 6 7 11 12 14 15) [ 0.289089] ACPI: PCI Interrupt Link [LNKC] (IRQs 1 3 *4 5 6 7 10 12 14 15) [ 0.289350] ACPI: PCI Interrupt Link [LNKD] (IRQs 1 *3 4 5 6 7 11 12 14 15) [ 0.289606] ACPI: PCI Interrupt Link [LNKE] (IRQs 1 3 4 5 6 7 10 12 14 15) *0, disabled. [ 0.289861] ACPI: PCI Interrupt Link [LNKF] (IRQs 1 3 4 5 6 7 11 12 14 15) *0, disabled. [ 0.290128] ACPI: PCI Interrupt Link [LNKG] (IRQs 1 3 4 5 6 7 10 12 14 15) *11 [ 0.290389] ACPI: PCI Interrupt Link [LNKH] (IRQs 1 3 4 5 6 *7 11 12 14 15) [ 0.290937] ACPI: Enabled 3 GPEs in block 00 to 1F [ 0.290957] ACPI: \_SB_.PCI0: notify handler is installed [ 0.291065] Found 1 acpi root devices [ 0.291266] vgaarb: device added: PCI:0000:01:00.0,decodes=io+mem,owns=io+mem,locks=none [ 0.291272] vgaarb: loaded [ 0.291275] vgaarb: bridge control possible 0000:01:00.0 [ 0.291535] SCSI subsystem initialized [ 0.291541] ACPI: bus type ATA registered [ 0.291682] libata version 3.00 loaded. [ 0.291701] ACPI: bus type USB registered [ 0.291762] usbcore: registered new interface driver usbfs [ 0.291797] usbcore: registered new interface driver hub [ 0.291875] usbcore: registered new device driver usb [ 0.292087] PCI: Using ACPI for IRQ routing [ 0.304952] PCI: pci_cache_line_size set to 64 bytes [ 0.305047] e820: reserve RAM buffer [mem 0x0009fc00-0x0009ffff] [ 0.305053] e820: reserve RAM buffer [mem 0xbf7f4000-0xbfffffff] [ 0.305059] e820: reserve RAM buffer [mem 0xbf8ad000-0xbfffffff] [ 0.305063] e820: reserve RAM buffer [mem 0xbf8cd000-0xbfffffff] [ 0.305068] e820: reserve RAM buffer [mem 0xbfd1b000-0xbfffffff] [ 0.305072] e820: reserve RAM buffer [mem 0xbfef0000-0xbfffffff] [ 0.305297] NetLabel: Initializing [ 0.305302] NetLabel: domain hash size = 128 [ 0.305304] NetLabel: protocols = UNLABELED CIPSOv4 [ 0.305332] NetLabel: unlabeled traffic allowed by default [ 0.305430] hpet0: at MMIO 0xfed00000, IRQs 2, 8, 0 [ 0.305439] hpet0: 3 comparators, 64-bit 14.318180 MHz counter [ 0.307505] Switched to clocksource hpet [ 0.313499] AppArmor: AppArmor Filesystem Enabled [ 0.313595] pnp: PnP ACPI init [ 0.313626] ACPI: bus type PNP registered [ 0.313714] pnp 00:00: [dma 4] [ 0.313775] pnp 00:00: Plug and Play ACPI device, IDs PNP0200 (active) [ 0.313852] pnp 00:01: Plug and Play ACPI device, IDs INT0800 (active) [ 0.314234] system 00:02: [mem 0xfed00000-0xfed003ff] has been reserved [ 0.314243] system 00:02: Plug and Play ACPI device, IDs PNP0103 PNP0c01 (active) [ 0.314344] pnp 00:03: Plug and Play ACPI device, IDs PNP0c04 (active) [ 0.314465] system 00:04: [io 0x0680-0x069f] has been reserved [ 0.314472] system 00:04: [io 0x1000-0x100f] has been reserved [ 0.314478] system 00:04: [io 0xffff] has been reserved [ 0.314484] system 00:04: [io 0xffff] has been reserved [ 0.314491] system 00:04: [io 0x0400-0x047f] could not be reserved [ 0.314497] system 00:04: [io 0x0500-0x057f] has been reserved [ 0.314531] system 00:04: [io 0x0600-0x061f] has been reserved [ 0.314539] system 00:04: Plug and Play ACPI device, IDs PNP0c02 (active) [ 0.314712] system 00:05: [io 0x06a0-0x06af] has been reserved [ 0.314719] system 00:05: [io 0x06b0-0x06ff] has been reserved [ 0.314726] system 00:05: Plug and Play ACPI device, IDs PNP0c02 (active) [ 0.314813] pnp 00:06: Plug and Play ACPI device, IDs PNP0b00 (active) [ 0.315318] system 00:07: [io 0x0a00-0x0a0f] has been reserved [ 0.315326] system 00:07: [io 0x0a10-0x0a1f] has been reserved [ 0.315333] system 00:07: Plug and Play ACPI device, IDs PNP0c02 (active) [ 0.315528] system 00:08: [io 0x04d0-0x04d1] has been reserved [ 0.315536] system 00:08: Plug and Play ACPI device, IDs PNP0c02 (active) [ 0.316026] system 00:09: [mem 0xfed1c000-0xfed1ffff] has been reserved [ 0.316034] system 00:09: [mem 0x00000000-0x00003fff] could not be reserved [ 0.316040] system 00:09: [mem 0x00000000-0x00000fff] could not be reserved [ 0.316046] system 00:09: [mem 0x00000000-0x00000fff] could not be reserved [ 0.316052] system 00:09: [mem 0xfed45000-0xfed8ffff] has been reserved [ 0.316059] system 00:09: Plug and Play ACPI device, IDs PNP0c02 (active) [ 0.316847] system 00:0a: [mem 0xfed14000-0xfed19fff] has been reserved [ 0.316855] system 00:0a: [mem 0xe0000000-0xefffffff] has been reserved [ 0.316862] system 00:0a: Plug and Play ACPI device, IDs PNP0c01 (active) [ 0.316948] pnp 00:0b: Plug and Play ACPI device, IDs INT0800 (active) [ 0.317650] system 00:0c: [io 0x0400-0x047f] could not be reserved [ 0.317658] system 00:0c: [io 0x0500-0x053f] has been reserved [ 0.317665] system 00:0c: [mem 0xfec00000-0xfec00fff] could not be reserved [ 0.317672] system 00:0c: [mem 0xfee00000-0xfee00fff] has been reserved [ 0.317678] system 00:0c: [mem 0xfed1c000-0xfed1ffff] has been reserved [ 0.317684] system 00:0c: [mem 0xfed20000-0xfed8ffff] could not be reserved [ 0.317690] system 00:0c: [mem 0xffe00000-0xffffffff] has been reserved [ 0.317697] system 00:0c: Plug and Play ACPI device, IDs PNP0c02 (active) [ 0.317707] pnp: PnP ACPI: found 13 devices [ 0.317710] ACPI: bus type PNP unregistered [ 0.328034] pci 0000:00:1c.0: bridge window [mem 0x00100000-0x000fffff 64bit pref] to [bus 01] add_size 200000 [ 0.328057] pci 0000:00:1c.1: bridge window [mem 0x00100000-0x000fffff 64bit pref] to [bus 02] add_size 200000 [ 0.328075] pci 0000:00:1c.3: bridge window [mem 0x00100000-0x000fffff 64bit pref] to [bus 03] add_size 200000 [ 0.328105] pci 0000:00:1c.0: res[15]=[mem 0x00100000-0x000fffff 64bit pref] get_res_add_size add_size 200000 [ 0.328112] pci 0000:00:1c.1: res[15]=[mem 0x00100000-0x000fffff 64bit pref] get_res_add_size add_size 200000 [ 0.328117] pci 0000:00:1c.3: res[15]=[mem 0x00100000-0x000fffff 64bit pref] get_res_add_size add_size 200000 [ 0.328130] pci 0000:00:1c.0: BAR 15: assigned [mem 0xd0400000-0xd05fffff 64bit pref] [ 0.328139] pci 0000:00:1c.1: BAR 15: assigned [mem 0xd0600000-0xd07fffff 64bit pref] [ 0.328147] pci 0000:00:1c.3: BAR 15: assigned [mem 0xd0800000-0xd09fffff 64bit pref] [ 0.328154] pci 0000:00:1c.0: PCI bridge to [bus 01] [ 0.328162] pci 0000:00:1c.0: bridge window [io 0xe000-0xefff] [ 0.328173] pci 0000:00:1c.0: bridge window [mem 0xc0000000-0xd00fffff] [ 0.328182] pci 0000:00:1c.0: bridge window [mem 0xd0400000-0xd05fffff 64bit pref] [ 0.328194] pci 0000:00:1c.1: PCI bridge to [bus 02] [ 0.328201] pci 0000:00:1c.1: bridge window [io 0xd000-0xdfff] [ 0.328211] pci 0000:00:1c.1: bridge window [mem 0xd0200000-0xd02fffff] [ 0.328219] pci 0000:00:1c.1: bridge window [mem 0xd0600000-0xd07fffff 64bit pref] [ 0.328231] pci 0000:00:1c.3: PCI bridge to [bus 03] [ 0.328238] pci 0000:00:1c.3: bridge window [io 0xc000-0xcfff] [ 0.328248] pci 0000:00:1c.3: bridge window [mem 0xd0100000-0xd01fffff] [ 0.328256] pci 0000:00:1c.3: bridge window [mem 0xd0800000-0xd09fffff 64bit pref] [ 0.328269] pci 0000:00:1e.0: PCI bridge to [bus 04] [ 0.328746] pci 0000:00:1e.0: setting latency timer to 64 [ 0.328756] pci_bus 0000:00: resource 4 [io 0x0000-0x0cf7] [ 0.328761] pci_bus 0000:00: resource 5 [io 0x0d00-0xffff] [ 0.328766] pci_bus 0000:00: resource 6 [mem 0x000a0000-0x000bffff] [ 0.328771] pci_bus 0000:00: resource 7 [mem 0x000c0000-0x000dffff] [ 0.328776] pci_bus 0000:00: resource 8 [mem 0x000e0000-0x000effff] [ 0.328781] pci_bus 0000:00: resource 9 [mem 0x000f0000-0x000fffff] [ 0.328786] pci_bus 0000:00: resource 10 [mem 0xc0000000-0xfebfffff] [ 0.328791] pci_bus 0000:01: resource 0 [io 0xe000-0xefff] [ 0.328796] pci_bus 0000:01: resource 1 [mem 0xc0000000-0xd00fffff] [ 0.328801] pci_bus 0000:01: resource 2 [mem 0xd0400000-0xd05fffff 64bit pref] [ 0.328806] pci_bus 0000:02: resource 0 [io 0xd000-0xdfff] [ 0.328811] pci_bus 0000:02: resource 1 [mem 0xd0200000-0xd02fffff] [ 0.328816] pci_bus 0000:02: resource 2 [mem 0xd0600000-0xd07fffff 64bit pref] [ 0.328822] pci_bus 0000:03: resource 0 [io 0xc000-0xcfff] [ 0.328826] pci_bus 0000:03: resource 1 [mem 0xd0100000-0xd01fffff] [ 0.328832] pci_bus 0000:03: resource 2 [mem 0xd0800000-0xd09fffff 64bit pref] [ 0.328837] pci_bus 0000:04: resource 4 [io 0x0000-0x0cf7] [ 0.328842] pci_bus 0000:04: resource 5 [io 0x0d00-0xffff] [ 0.328847] pci_bus 0000:04: resource 6 [mem 0x000a0000-0x000bffff] [ 0.328851] pci_bus 0000:04: resource 7 [mem 0x000c0000-0x000dffff] [ 0.328856] pci_bus 0000:04: resource 8 [mem 0x000e0000-0x000effff] [ 0.328861] pci_bus 0000:04: resource 9 [mem 0x000f0000-0x000fffff] [ 0.328866] pci_bus 0000:04: resource 10 [mem 0xc0000000-0xfebfffff] [ 0.329013] NET: Registered protocol family 2 [ 0.329579] TCP established hash table entries: 32768 (order: 7, 524288 bytes) [ 0.329949] TCP bind hash table entries: 32768 (order: 7, 524288 bytes) [ 0.330267] TCP: Hash tables configured (established 32768 bind 32768) [ 0.330390] TCP: reno registered [ 0.330420] UDP hash table entries: 2048 (order: 4, 65536 bytes) [ 0.330483] UDP-Lite hash table entries: 2048 (order: 4, 65536 bytes) [ 0.330773] NET: Registered protocol family 1 [ 0.332266] pci 0000:01:00.0: Boot video device [ 0.332287] PCI: CLS 64 bytes, default 64 [ 0.332425] Unpacking initramfs... [ 0.907296] Freeing initrd memory: 25092K (ffff880034eee000 - ffff88003676f000) [ 0.907307] PCI-DMA: Using software bounce buffering for IO (SWIOTLB) [ 0.907313] software IO TLB [mem 0xbb7f4000-0xbf7f4000] (64MB) mapped at [ffff8800bb7f4000-ffff8800bf7f3fff] [ 0.908255] microcode: CPU0 sig=0x30661, pf=0x4, revision=0x10d [ 0.908271] microcode: CPU1 sig=0x30661, pf=0x4, revision=0x10d [ 0.908284] microcode: CPU2 sig=0x30661, pf=0x4, revision=0x10d [ 0.908301] microcode: CPU3 sig=0x30661, pf=0x4, revision=0x10d [ 0.908391] microcode: Microcode Update Driver: v2.00 <tigran(a)aivazian.fsnet.co.uk>, Peter Oruba [ 0.908396] Scanning for low memory corruption every 60 seconds [ 0.909133] audit: initializing netlink socket (disabled) [ 0.909169] type=2000 audit(1379599809.803:1): initialized [ 0.948091] HugeTLB registered 2 MB page size, pre-allocated 0 pages [ 0.948801] zbud: loaded [ 0.949199] VFS: Disk quotas dquot_6.5.2 [ 0.949263] Dquot-cache hash table entries: 512 (order 0, 4096 bytes) [ 0.949786] msgmni has been set to 7908 [ 0.950392] Block layer SCSI generic (bsg) driver version 0.4 loaded (major 252) [ 0.950484] io scheduler noop registered [ 0.950490] io scheduler deadline registered [ 0.950533] io scheduler cfq registered (default) [ 0.950742] pcieport 0000:00:1c.0: irq 40 for MSI/MSI-X [ 0.950900] pcieport 0000:00:1c.1: irq 41 for MSI/MSI-X [ 0.951046] pcieport 0000:00:1c.3: irq 42 for MSI/MSI-X [ 0.951216] pci_hotplug: PCI Hot Plug PCI Core version: 0.5 [ 0.951253] pciehp: PCI Express Hot Plug Controller Driver version: 0.4 [ 0.951366] vesafb: mode is 1400x1050x32, linelength=5632, pages=0 [ 0.951369] vesafb: scrolling: redraw [ 0.951374] vesafb: Truecolor: size=0:8:8:8, shift=0:16:8:0 [ 0.952335] vesafb: framebuffer at 0xc0000000, mapped to 0xffffc90010800000, using 5824k, total 5824k [ 1.742876] Console: switching to colour frame buffer device 175x65 [ 1.910348] tsc: Refined TSC clocksource calibration: 1861.999 MHz [ 2.529563] fb0: VESA VGA frame buffer device [ 2.529598] intel_idle: does not run on family 6 model 54 [ 2.529650] GHES: HEST is not enabled! [ 2.529840] Serial: 8250/16550 driver, 32 ports, IRQ sharing disabled [ 2.534053] Non-volatile memory driver v1.3 [ 2.534059] Linux agpgart interface v0.103 [ 2.534365] ahci 0000:00:1f.2: version 3.0 [ 2.534586] ahci 0000:00:1f.2: irq 43 for MSI/MSI-X [ 2.534656] ahci: SSS flag set, parallel bus scan disabled [ 2.534708] ahci 0000:00:1f.2: AHCI 0001.0100 32 slots 4 ports 3 Gbps 0x3 impl SATA mode [ 2.534715] ahci 0000:00:1f.2: flags: 64bit ncq stag pm led clo pio slum part [ 2.534753] ahci 0000:00:1f.2: setting latency timer to 64 [ 2.535805] scsi0 : ahci [ 2.536050] scsi1 : ahci [ 2.536220] scsi2 : ahci [ 2.536396] scsi3 : ahci [ 2.536506] ata1: SATA max UDMA/133 abar m1024@0xd0304000 port 0xd0304100 irq 43 [ 2.536513] ata2: SATA max UDMA/133 abar m1024@0xd0304000 port 0xd0304180 irq 43 [ 2.536517] ata3: DUMMY [ 2.536520] ata4: DUMMY [ 2.536591] ehci_hcd: USB 2.0 'Enhanced' Host Controller (EHCI) Driver [ 2.536597] ehci-pci: EHCI PCI platform driver [ 2.536889] ehci-pci 0000:00:1d.7: setting latency timer to 64 [ 2.536912] ehci-pci 0000:00:1d.7: EHCI Host Controller [ 2.536933] ehci-pci 0000:00:1d.7: new USB bus registered, assigned bus number 1 [ 2.536972] ehci-pci 0000:00:1d.7: debug port 1 [ 2.540913] ehci-pci 0000:00:1d.7: cache line size of 64 is not supported [ 2.540959] ehci-pci 0000:00:1d.7: irq 23, io mem 0xd0305000 [ 2.546747] ehci-pci 0000:00:1d.7: USB 2.0 started, EHCI 1.00 [ 2.546799] usb usb1: New USB device found, idVendor=1d6b, idProduct=0002 [ 2.546805] usb usb1: New USB device strings: Mfr=3, Product=2, SerialNumber=1 [ 2.546809] usb usb1: Product: EHCI Host Controller [ 2.546814] usb usb1: Manufacturer: Linux 3.11.1-1.g1383321-desktop ehci_hcd [ 2.546818] usb usb1: SerialNumber: 0000:00:1d.7 [ 2.547119] hub 1-0:1.0: USB hub found [ 2.547131] hub 1-0:1.0: 8 ports detected [ 2.547892] ohci_hcd: USB 1.1 'Open' Host Controller (OHCI) Driver [ 2.547896] uhci_hcd: USB Universal Host Controller Interface driver [ 2.548104] uhci_hcd 0000:00:1d.0: setting latency timer to 64 [ 2.548112] uhci_hcd 0000:00:1d.0: UHCI Host Controller [ 2.548129] uhci_hcd 0000:00:1d.0: new USB bus registered, assigned bus number 2 [ 2.548176] uhci_hcd 0000:00:1d.0: irq 23, io base 0x0000f0a0 [ 2.548262] usb usb2: New USB device found, idVendor=1d6b, idProduct=0001 [ 2.548267] usb usb2: New USB device strings: Mfr=3, Product=2, SerialNumber=1 [ 2.548272] usb usb2: Product: UHCI Host Controller [ 2.548277] usb usb2: Manufacturer: Linux 3.11.1-1.g1383321-desktop uhci_hcd [ 2.548281] usb usb2: SerialNumber: 0000:00:1d.0 [ 2.548575] hub 2-0:1.0: USB hub found [ 2.548591] hub 2-0:1.0: 2 ports detected [ 2.549013] uhci_hcd 0000:00:1d.1: setting latency timer to 64 [ 2.549021] uhci_hcd 0000:00:1d.1: UHCI Host Controller [ 2.549033] uhci_hcd 0000:00:1d.1: new USB bus registered, assigned bus number 3 [ 2.549105] uhci_hcd 0000:00:1d.1: irq 19, io base 0x0000f080 [ 2.549191] usb usb3: New USB device found, idVendor=1d6b, idProduct=0001 [ 2.549196] usb usb3: New USB device strings: Mfr=3, Product=2, SerialNumber=1 [ 2.549201] usb usb3: Product: UHCI Host Controller [ 2.549206] usb usb3: Manufacturer: Linux 3.11.1-1.g1383321-desktop uhci_hcd [ 2.549210] usb usb3: SerialNumber: 0000:00:1d.1 [ 2.549483] hub 3-0:1.0: USB hub found [ 2.549497] hub 3-0:1.0: 2 ports detected [ 2.549903] uhci_hcd 0000:00:1d.2: setting latency timer to 64 [ 2.549911] uhci_hcd 0000:00:1d.2: UHCI Host Controller [ 2.549923] uhci_hcd 0000:00:1d.2: new USB bus registered, assigned bus number 4 [ 2.549997] uhci_hcd 0000:00:1d.2: irq 18, io base 0x0000f060 [ 2.550081] usb usb4: New USB device found, idVendor=1d6b, idProduct=0001 [ 2.550086] usb usb4: New USB device strings: Mfr=3, Product=2, SerialNumber=1 [ 2.550091] usb usb4: Product: UHCI Host Controller [ 2.550096] usb usb4: Manufacturer: Linux 3.11.1-1.g1383321-desktop uhci_hcd [ 2.550100] usb usb4: SerialNumber: 0000:00:1d.2 [ 2.550378] hub 4-0:1.0: USB hub found [ 2.550392] hub 4-0:1.0: 2 ports detected [ 2.550796] uhci_hcd 0000:00:1d.3: setting latency timer to 64 [ 2.550803] uhci_hcd 0000:00:1d.3: UHCI Host Controller [ 2.550824] uhci_hcd 0000:00:1d.3: new USB bus registered, assigned bus number 5 [ 2.550895] uhci_hcd 0000:00:1d.3: irq 16, io base 0x0000f040 [ 2.550979] usb usb5: New USB device found, idVendor=1d6b, idProduct=0001 [ 2.550985] usb usb5: New USB device strings: Mfr=3, Product=2, SerialNumber=1 [ 2.550989] usb usb5: Product: UHCI Host Controller [ 2.550994] usb usb5: Manufacturer: Linux 3.11.1-1.g1383321-desktop uhci_hcd [ 2.550998] usb usb5: SerialNumber: 0000:00:1d.3 [ 2.551277] hub 5-0:1.0: USB hub found [ 2.551291] hub 5-0:1.0: 2 ports detected [ 2.551629] i8042: PNP: No PS/2 controller found. Probing ports directly. [ 2.840811] ata1: SATA link up 3.0 Gbps (SStatus 123 SControl 300) [ 3.077348] Switched to clocksource tsc [ 3.602621] ata1.00: ATA-8: KINGSTON SV300S37A60G, 505ABBF0, max UDMA/133 [ 3.602627] ata1.00: 117231408 sectors, multi 16: LBA48 NCQ (depth 31/32), AA [ 3.603675] i8042: No controller found [ 3.606784] mousedev: PS/2 mouse device common for all mice [ 3.607032] rtc_cmos 00:06: RTC can wake from S4 [ 3.607277] rtc_cmos 00:06: rtc core: registered rtc_cmos as rtc0 [ 3.607328] rtc_cmos 00:06: alarms up to one month, y3k, 242 bytes nvram, hpet irqs [ 3.607349] cpuidle: using governor ladder [ 3.607353] cpuidle: using governor menu [ 3.607361] ledtrig-cpu: registered to indicate activity on CPUs [ 3.607399] hidraw: raw HID events driver (C) Jiri Kosina [ 3.607611] usbcore: registered new interface driver usbhid [ 3.607615] usbhid: USB HID core driver [ 3.607799] TCP: cubic registered [ 3.607907] NET: Registered protocol family 10 [ 3.608294] Key type dns_resolver registered [ 3.608976] PM: Checking hibernation image partition /dev/disk/by-id/ata-KINGSTON_SV300S37A60G_50026B723106E9DA-part1 [ 3.614537] ata1.00: configured for UDMA/133 [ 3.614794] scsi 0:0:0:0: Direct-Access ATA KINGSTON SV300S3 505A PQ: 0 ANSI: 5 [ 3.615223] sd 0:0:0:0: [sda] 117231408 512-byte logical blocks: (60.0 GB/55.8 GiB) [ 3.615484] sd 0:0:0:0: [sda] Write Protect is off [ 3.615492] sd 0:0:0:0: [sda] Mode Sense: 00 3a 00 00 [ 3.615566] sd 0:0:0:0: [sda] Write cache: enabled, read cache: enabled, doesn't support DPO or FUA [ 3.616644] sda: sda1 sda2 sda3 [ 3.617504] sd 0:0:0:0: [sda] Attached SCSI disk [ 3.919889] ata2: SATA link down (SStatus 0 SControl 300) [ 3.919995] PM: Hibernation image not present or could not be loaded. [ 3.920049] registered taskstats version 1 [ 3.921162] Magic number: 13:546:182 [ 3.921351] rtc_cmos 00:06: setting system clock to 2013-09-19 14:10:13 UTC (1379599813) [ 3.924713] Freeing unused kernel memory: 1332K (ffffffff81cc4000 - ffffffff81e11000) [ 3.924721] Write protecting the kernel read-only data: 12288k [ 3.926920] Freeing unused kernel memory: 264K (ffff8800015be000 - ffff880001600000) [ 3.938318] Freeing unused kernel memory: 1776K (ffff880001a44000 - ffff880001c00000) [ 4.048266] alua: device handler registered [ 4.051640] emc: device handler registered [ 4.054978] hp_sw: device handler registered [ 4.058430] rdac: device handler registered [ 4.059898] usb 2-1: new full-speed USB device number 2 using uhci_hcd [ 4.070181] systemd-udevd[131]: starting version 207 [ 4.134183] thermal LNXTHERM:00: registered as thermal_zone0 [ 4.134196] ACPI: Thermal Zone [TZ01] (27 C) [ 4.134482] ACPI: Fan [FAN0] (off) [ 4.141405] [drm] Initialized drm 1.1.0 20060810 [ 4.156991] [drm] radeon kernel modesetting enabled. [ 4.157158] checking generic (c0000000 5b0000) vs hw (c0000000 10000000) [ 4.157166] fb: conflicting fb hw usage radeondrmfb vs VESA VGA - removing generic driver [ 4.157211] Console: switching to colour dummy device 80x25 [ 4.158551] [drm] initializing kernel modesetting (CAICOS 0x1002:0x6761 0x1297:0x4012). [ 4.158615] [drm] register mmio base: 0xD0020000 [ 4.158621] [drm] register mmio size: 131072 [ 4.158976] ATOM BIOS: Shuttle [ 4.159598] radeon 0000:01:00.0: VRAM: 512M 0x0000000000000000 - 0x000000001FFFFFFF (512M used) [ 4.159609] radeon 0000:01:00.0: GTT: 512M 0x0000000020000000 - 0x000000003FFFFFFF [ 4.159614] [drm] Detected VRAM RAM=512M, BAR=256M [ 4.159618] [drm] RAM width 64bits DDR [ 4.159801] [TTM] Zone kernel: Available graphics memory: 2026338 kiB [ 4.159809] [TTM] Initializing pool allocator [ 4.159823] [TTM] Initializing DMA pool allocator [ 4.159954] [drm] radeon: 512M of VRAM memory ready [ 4.159960] [drm] radeon: 512M of GTT memory ready. [ 4.160922] [drm] GART: num cpu pages 131072, num gpu pages 131072 [ 4.171257] [drm] Loading CAICOS Microcode [ 4.181153] [drm] PCIE GART of 512M enabled (table at 0x0000000000273000). [ 4.181584] radeon 0000:01:00.0: WB enabled [ 4.181597] radeon 0000:01:00.0: fence driver on ring 0 use gpu addr 0x0000000020000c00 and cpu addr 0xffff880135e03c00 [ 4.181607] radeon 0000:01:00.0: fence driver on ring 3 use gpu addr 0x0000000020000c0c and cpu addr 0xffff880135e03c0c [ 4.200521] radeon 0000:01:00.0: fence driver on ring 5 use gpu addr 0x0000000000072118 and cpu addr 0xffffc90010eb2118 [ 4.200534] [drm] Supports vblank timestamp caching Rev 1 (10.10.2010). [ 4.200539] [drm] Driver supports precise vblank timestamp query. [ 4.200611] radeon 0000:01:00.0: irq 44 for MSI/MSI-X [ 4.200650] radeon 0000:01:00.0: radeon: using MSI. [ 4.200746] [drm] radeon: irq initialized. [ 4.214436] usb 2-1: New USB device found, idVendor=045e, idProduct=0745 [ 4.214446] usb 2-1: New USB device strings: Mfr=1, Product=2, SerialNumber=0 [ 4.214454] usb 2-1: Product: Microsoft® 2.4GHz Transceiver v7.0 [ 4.214461] usb 2-1: Manufacturer: Microsoft [ 4.222120] input: Microsoft Microsoft® 2.4GHz Transceiver v7.0 as /devices/pci0000:00/0000:00:1d.0/usb2/2-1/2-1:1.0/input/input0 [ 4.222490] hid-generic 0003:045E:0745.0001: input,hidraw0: USB HID v1.11 Keyboard [Microsoft Microsoft® 2.4GHz Transceiver v7.0] on usb-0000:00:1d.0-1/input0 [ 4.222815] [drm] ring test on 0 succeeded in 1 usecs [ 4.222930] [drm] ring test on 3 succeeded in 1 usecs [ 4.229641] input: Microsoft Microsoft® 2.4GHz Transceiver v7.0 as /devices/pci0000:00/0000:00:1d.0/usb2/2-1/2-1:1.1/input/input1 [ 4.230095] hid-generic 0003:045E:0745.0002: input,hidraw1: USB HID v1.11 Mouse [Microsoft Microsoft® 2.4GHz Transceiver v7.0] on usb-0000:00:1d.0-1/input1 [ 4.255991] input: Microsoft Microsoft® 2.4GHz Transceiver v7.0 as /devices/pci0000:00/0000:00:1d.0/usb2/2-1/2-1:1.2/input/input2 [ 4.256486] hid-generic 0003:045E:0745.0003: input,hiddev0,hidraw2: USB HID v1.11 Device [Microsoft Microsoft® 2.4GHz Transceiver v7.0] on usb-0000:00:1d.0-1/input2 [ 4.280722] xor: measuring software checksum speed [ 4.289905] prefetch64-sse: 4540.000 MB/sec [ 4.301956] generic_sse: 4224.000 MB/sec [ 4.301964] xor: using function: prefetch64-sse (4540.000 MB/sec) [ 4.320256] raid6: sse2x1 156 MB/s [ 4.336997] raid6: sse2x2 257 MB/s [ 4.353993] raid6: sse2x4 523 MB/s [ 4.353999] raid6: using algorithm sse2x4 (523 MB/s) [ 4.354004] raid6: using ssse3x2 recovery algorithm [ 4.361309] bio: create slab <bio-1> at 1 [ 4.361660] Btrfs loaded [ 4.412706] [drm] ring test on 5 succeeded in 1 usecs [ 4.412719] [drm] UVD initialized successfully. [ 4.413723] [drm] ib test on ring 0 succeeded in 0 usecs [ 4.413845] [drm] ib test on ring 3 succeeded in 0 usecs [ 4.566733] [drm] ib test on ring 5 succeeded [ 4.567346] [drm] Radeon Display Connectors [ 4.567350] [drm] Connector 0: [ 4.567353] [drm] HDMI-A-1 [ 4.567356] [drm] HPD1 [ 4.567361] [drm] DDC: 0x6430 0x6430 0x6434 0x6434 0x6438 0x6438 0x643c 0x643c [ 4.567364] [drm] Encoders: [ 4.567367] [drm] DFP1: INTERNAL_UNIPHY1 [ 4.567370] [drm] Connector 1: [ 4.567373] [drm] VGA-1 [ 4.567377] [drm] DDC: 0x64d8 0x64d8 0x64dc 0x64dc 0x64e0 0x64e0 0x64e4 0x64e4 [ 4.567379] [drm] Encoders: [ 4.567382] [drm] CRT1: INTERNAL_KLDSCP_DAC1 [ 4.567541] [drm] Internal thermal controller with fan control [ 4.568762] [drm] radeon: power management initialized [ 4.710870] [drm] fb mappable at 0xC0375000 [ 4.710874] [drm] vram apper at 0xC0000000 [ 4.710877] [drm] size 8294400 [ 4.710880] [drm] fb depth is 24 [ 4.710883] [drm] pitch is 7680 [ 4.711041] fbcon: radeondrmfb (fb0) is primary device [ 4.932297] Console: switching to colour frame buffer device 240x67 [ 4.999732] radeon 0000:01:00.0: fb0: radeondrmfb frame buffer device [ 4.999737] radeon 0000:01:00.0: registered panic notifier [ 4.999768] [drm] Initialized radeon 2.34.0 20080528 for 0000:01:00.0 on minor 0 [ 7.713893] NET: Registered protocol family 17 [ 7.759016] PM: Marking nosave pages: [mem 0x0009f000-0x000fffff] [ 7.759029] PM: Marking nosave pages: [mem 0xbf7f4000-0xbf8abfff] [ 7.759040] PM: Marking nosave pages: [mem 0xbf8ad000-0xbf8cafff] [ 7.759044] PM: Marking nosave pages: [mem 0xbf8cd000-0xbf944fff] [ 7.759052] PM: Marking nosave pages: [mem 0xbfd1b000-0xbfee4fff] [ 7.759071] PM: Marking nosave pages: [mem 0xbfef0000-0xffffffff] [ 7.761713] PM: Basic memory bitmaps created [ 7.816156] PM: Basic memory bitmaps freed [ 7.876380] PM: Starting manual resume from disk [ 7.876389] PM: Hibernation image partition 8:1 present [ 7.876393] PM: Looking for hibernation image. [ 7.877975] PM: Image not found (code -22) [ 7.877980] PM: Hibernation image not present or could not be loaded. [ 7.931129] btrfs: disk space caching is enabled [ 7.938125] Btrfs detected SSD devices, enabling SSD mode [ 8.194664] device-mapper: uevent: version 1.0.3 [ 8.194857] device-mapper: ioctl: 4.25.0-ioctl (2013-06-26) initialised: dm-devel(a)redhat.com [ 8.197390] LVM: Activation generator successfully completed. [ 8.581758] sd 0:0:0:0: Attached scsi generic sg0 type 0 [ 8.594831] systemd-udevd[330]: starting version 207 [ 8.722650] btrfs: disk space caching is enabled [ 8.849759] input: Power Button as /devices/LNXSYSTM:00/device:00/PNP0C0C:00/input/input3 [ 8.849917] ACPI: Power Button [PWRB] [ 8.850086] input: Sleep Button as /devices/LNXSYSTM:00/device:00/PNP0C0E:00/input/input4 [ 8.850235] ACPI: Sleep Button [SLPB] [ 8.850447] input: Power Button as /devices/LNXSYSTM:00/LNXPWRBN:00/input/input5 [ 8.850633] ACPI: Power Button [PWRF] [ 8.917055] shpchp: Standard Hot Plug PCI Controller Driver version: 0.4 [ 8.987771] r8169 Gigabit Ethernet driver 2.3LK-NAPI loaded [ 8.987813] r8169 0000:02:00.2: can't disable ASPM; OS doesn't have ASPM control [ 8.988222] r8169 0000:02:00.2: irq 45 for MSI/MSI-X [ 8.988914] r8169 0000:02:00.2 eth0: RTL8411 at 0xffffc90000616000, 80:ee:73:5d:59:d5, XID 08800800 IRQ 45 [ 8.988926] r8169 0000:02:00.2 eth0: jumbo features [frames: 9200 bytes, tx checksumming: ko] [ 8.989473] rtsx_pci 0000:02:00.0: irq 46 for MSI/MSI-X [ 8.989519] rtsx_pci 0000:02:00.0: rtsx_pci_acquire_irq: pcr->msi_en = 1, pci->irq = 46 [ 9.042648] i801_smbus 0000:00:1f.3: SMBus using PCI Interrupt [ 9.045703] ACPI Warning: 0x0000000000000428-0x000000000000042f SystemIO conflicts with Region \SWC1 1 (20130517/utaddress-251) [ 9.045723] ACPI Warning: 0x0000000000000428-0x000000000000042f SystemIO conflicts with Region \PMIO 2 (20130517/utaddress-251) [ 9.045735] ACPI: If an ACPI driver is available for this device, you should use it instead of the native driver [ 9.045744] ACPI Warning: 0x0000000000000530-0x000000000000053f SystemIO conflicts with Region \GPIO 1 (20130517/utaddress-251) [ 9.045755] ACPI: If an ACPI driver is available for this device, you should use it instead of the native driver [ 9.045761] ACPI Warning: 0x0000000000000500-0x000000000000052f SystemIO conflicts with Region \GPIO 1 (20130517/utaddress-251) [ 9.045771] ACPI: If an ACPI driver is available for this device, you should use it instead of the native driver [ 9.045775] lpc_ich: Resource conflict(s) found affecting gpio_ich [ 9.069532] input: PC Speaker as /devices/platform/pcspkr/input/input6 [ 9.161452] snd_hda_intel 0000:00:1b.0: irq 47 for MSI/MSI-X [ 9.173414] coretemp coretemp.0: Unable to read TjMax from CPU 0 [ 9.173454] coretemp coretemp.0: Unable to read TjMax from CPU 2 [ 9.183151] ALSA hda_auto_parser.c:393 autoconfig: line_outs=1 (0xd/0x0/0x0/0x0/0x0) type:speaker [ 9.183163] ALSA hda_auto_parser.c:397 speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 9.183169] ALSA hda_auto_parser.c:401 hp_outs=1 (0xb/0x0/0x0/0x0/0x0) [ 9.183175] ALSA hda_auto_parser.c:402 mono: mono_out=0x0 [ 9.183179] ALSA hda_auto_parser.c:406 inputs: [ 9.183186] ALSA hda_auto_parser.c:410 Mic=0xa [ 9.185481] input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:1b.0/input/input7 [ 9.212908] input: HDA Intel HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:1b.0/sound/card0/input8 [ 9.213158] input: HDA Intel Front Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card0/input9 [ 9.213947] input: HDA Intel Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input10 [ 9.215296] ALSA hda_intel.c:3116 0000:01:00.1: Handle VGA-switcheroo audio client [ 9.215308] ALSA hda_intel.c:3317 0000:01:00.1: Using LPIB position fix [ 9.215809] snd_hda_intel 0000:01:00.1: irq 48 for MSI/MSI-X [ 9.219577] ALSA hda_intel.c:1787 0000:01:00.1: Enable sync_write for stable communication [ 9.221420] input: HD-Audio Generic HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:1c.0/0000:01:00.1/sound/card1/input11 [ 9.482902] iTCO_vendor_support: vendor-support=0 [ 9.484128] iTCO_wdt: Intel TCO WatchDog Timer Driver v1.10 [ 9.484199] iTCO_wdt: Found a NM10 TCO device (Version=2, TCOBASE=0x0460) [ 9.485279] iTCO_wdt: initialized. heartbeat=30 sec (nowayout=0) [ 13.532853] r8169 0000:02:00.2 ens1f2: link down [ 13.533034] IPv6: ADDRCONF(NETDEV_UP): ens1f2: link is not ready [ 15.100806] r8169 0000:02:00.2 ens1f2: link up [ 15.100824] IPv6: ADDRCONF(NETDEV_CHANGE): ens1f2: link becomes ready [ 37.401917] fuse init (API version 7.22)

2 4

[alsa-devel] [RFC PATCH] ALSA : hda - not use assigned converters for all unused pins
by mengdong.lin＠intel.com 20 Sep '13

20 Sep '13

From: Mengdong Lin <mengdong.lin(a)intel.com> BIOS can mark a pin as "no physical connection" if the port is used by an integrated display which is not audio capable. And audio driver will overlook such pins. On Haswell, such a disconneted pin will keep muted and connected to the 1st converter by default. But if the 1st convertor is assigned to a connected pin for audio streaming. The muted disconnected pin can make the connected pin no sound output. So this patch avoids using assigned converters for all unused pins for Haswell, including the disconected pins. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 3d8cd044..4960695 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1152,29 +1152,34 @@ static void haswell_config_cvts(struct hda_codec *codec, int pin_id, int mux_id) { struct hdmi_spec *spec = codec->spec; - struct hdmi_spec_per_pin *per_pin; - int pin_idx, mux_idx; + hda_nid_t pin_nid; + int pin_idx, cvt_idx; int curr; - int err; - - for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { - per_pin = get_pin(spec, pin_idx); + struct hdmi_spec_per_cvt *per_cvt; + /* configure all 3 pins, including "no physical connection" ones */ + for (pin_nid = 5; pin_nid < 8; pin_nid++) { + pin_idx = pin_nid_to_pin_index(spec, pin_nid); if (pin_idx == pin_id) continue; - curr = snd_hda_codec_read(codec, per_pin->pin_nid, 0, + curr = snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_CONNECT_SEL, 0); + if (curr != mux_id) + continue; + + for (cvt_idx = 0; cvt_idx < spec->num_cvts; cvt_idx++) { + per_cvt = get_cvt(spec, cvt_idx); - /* Choose another unused converter */ - if (curr == mux_id) { - err = hdmi_choose_cvt(codec, pin_idx, NULL, &mux_idx); - if (err < 0) - return; - snd_printdd("HDMI: choose converter %d for pin %d\n", mux_idx, pin_idx); - snd_hda_codec_write_cache(codec, per_pin->pin_nid, 0, + /* Must not already be assigned */ + if (!per_cvt->assigned) { + snd_printdd("choose converter %d for pin %d\n", + cvt_idx, pin_idx); + snd_hda_codec_write_cache(codec, pin_nid, 0, AC_VERB_SET_CONNECT_SEL, - mux_idx); + cvt_idx); + break; + } } } } -- 1.8.1.2

3 4

Re: [alsa-devel] [PATCH 01/51] DMA-API: provide a helper to set both DMA and coherent DMA masks
by Russell King - ARM Linux 20 Sep '13

20 Sep '13

On Fri, Sep 20, 2013 at 02:21:37AM +0100, Ben Hutchings wrote: > On Thu, 2013-09-19 at 22:25 +0100, Russell King wrote: > [...] > > -dma_set_coherent_mask() will always be able to set the same or a > > -smaller mask as dma_set_mask(). However for the rare case that a > > +The coherent coherent mask will always be able to set the same or a > > +smaller mask as the streaming mask. However for the rare case that a > [...] > > The new wording doesn't make sense; a mask doesn't set itself. I would > suggest: > > "The coherent mask can always be set to the same or a smaller mask than > the streaming mask." Yes, the original sentence is not particularly good, but I think even your modified version can be interpreted as "a mask setting itself" for all the same reasons that the original can be (which mask does "same" refer to?) Even so, I prefer your version. Thanks. :)

1 0

[alsa-devel] [PATCH 1/3] ASoC: ab8500: Downgrade noisy log message
by Mark Brown 20 Sep '13

20 Sep '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/ab8500-codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 80555d7..2726e5a 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2606,7 +2606,7 @@ static int ab8500_codec_driver_probe(struct platform_device *pdev) static int ab8500_codec_driver_remove(struct platform_device *pdev) { - dev_info(&pdev->dev, "%s Enter.\n", __func__); + dev_dbg(&pdev->dev, "%s Enter.\n", __func__); snd_soc_unregister_codec(&pdev->dev); -- 1.8.4.rc3

2 3

[alsa-devel] Alsa mmap and asynchronous callbacks
by Bruce 20 Sep '13

20 Sep '13

Hi, I tried posting a query related to async callbacks on the alsa-user list, and the Hammerfall forum, but noone appeared to know what I was talking about. I'm hoping someone here can at least give me some clues. *Working:* I wrote an application, based around the alsa example code alsa-lib/test/pcm.c using mmap and asynchronous callback notifications. This application runs fine with 32 channels in and 2 out on a Hammerfall MADI system and under alsa 1.0.15. It now works on 2 different machines setup with SUSE 10, then with alsa (and card driver) upgraded to 1.0.15. alsamixer is used to setup the soundcard in 96kHz mode (double speed). /Note:/ For some reason (I can't recall) I never managed to get the playback async callback to work at the same time as capture callback, so I simply called the playback "top-up" routine from the capture async routine. That works fine. The application was used for some years ! *Problem:* SUSE 11 with alsa 1.0.18, same machine, same sound card, application hangs after a few callbacks. No more asynchronous callbacks occur after approx 1s of playback/record. However, the callbacks are working for a short time. alsamixer looks quite different for the Hammerfall with this newer alsa, and it does appear to be the change of alsa driver that causes the problem. SUSE 11/Ubuntu with alsa 1.0.27. I tried this configuration in order to try a different Xonar sound card. The application was modified for only 2 channels, and interleaved instead of non-interleaved sample format (as required by the hardware if not using plughw). Both Xonar and Hammerfall hang in the same way. aplay and alsamixer both work with both soundcards. The alsa calls I'm using do not appear to be in the deprecated list (except "snd_pcm_sw_params_set_xfer_align" in 1.0.27). Would you like me to post code fragments ? /Note: /As a workaround, I attempted to install the working 1.0.15 under the later kernal in Suse 11. Unfortunately ./configure fails in alsa-driver due to differing kernel hooks. It is not realistic to stay with the very old SUSE 10; much of the machine hardware is not recognised. -- Cheers, Bruce

1 0

[alsa-devel] No sound with Sony VAIO VPCZ1 (ALC889)
by Tormen 20 Sep '13

20 Sep '13

Hi, alsa-driver: 1.0.25.3453.g60df0 (the latest I guess -- I used alsa-compile.sh) Kernel: 3.9.6-1~bpo70+1 (debian 7.1 backport) Problem: Sound works fine, except that no speaker gets detected. The internal speakers in my Sony VAIO VPCZ1 notebook did get detected in the past though. If I am not mistaken, this is the problem: [ 5.864681] ALSA hda_auto_parser.c:397 speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) alsa-info.sh: !!################################ !!ALSA Information Script v 0.4.62 !!################################ !!Script ran on: Wed Jul 3 17:38:37 UTC 2013 !!Linux Distribution !!------------------ Debian GNU/Linux 7 \n \l PRETTY_NAME="Debian GNU/Linux 7 (wheezy)" NAME="Debian GNU/Linux" ID=debian HOME_URL="http://www.debian.org/" SUPPORT_URL="http://www.debian.org/support/" BUG_REPORT_URL="http://bugs.debian.org/" !!DMI Information !!--------------- Manufacturer: Sony Corporation Product Name: VPCZ1290S Product Version: J0049ZZ5 Firmware Version: R3030C3 !!Kernel Information !!------------------ Kernel release: 3.9-0.bpo.1-amd64 Operating System: GNU/Linux Architecture: x86_64 Processor: unknown SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.25.3453.g60df0 Library version: 1.0.25 Utilities version: 1.0.25 !!Loaded ALSA modules !!------------------- snd_hda_intel !!Sound Servers on this system !!---------------------------- Pulseaudio: Installed - Yes (/usr/bin/pulseaudio) Running - Yes !!Soundcards recognised by ALSA !!----------------------------- 0 [MID ]: HDA-Intel - HDA Intel MID HDA Intel MID at 0xb6420000 irq 45 !!PCI Soundcards installed in the system !!-------------------------------------- 00:1b.0 Audio device: Intel Corporation 5 Series/3400 Series Chipset High Definition Audio (rev 05) !!Advanced information - PCI Vendor/Device/Subsystem ID's !!------------------------------------------------------- 00:1b.0 0403: 8086:3b56 (rev 05) Subsystem: 104d:905a !!Modprobe options (Sound related) !!-------------------------------- snd_pcsp: index=-2 snd_usb_audio: index=-2 snd_atiixp_modem: index=-2 snd_intel8x0m: index=-2 snd_via82xx_modem: index=-2 snd_hda_intel: model=no-primary-hp !!Loaded sound module options !!--------------------------- !!Module: snd_hda_intel align_buffer_size : -1 bdl_pos_adj : 32,-1,-1,-1,-1,-1,-1,-1 beep_mode : Y,Y,Y,Y,Y,Y,Y,Y enable : Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : (null),(null),(null),(null),(null),(null),(null),(null) index : -1,-1,-1,-1,-1,-1,-1,-1 jackpoll_ms : 0,0,0,0,0,0,0,0 model : no-primary-hp,(null),(null),(null),(null),(null),(null),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null) position_fix : -1,-1,-1,-1,-1,-1,-1,-1 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1 probe_only : 0,0,0,0,0,0,0,0 single_cmd : N snoop : Y !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Realtek ALC889 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0889 Subsystem Id: 0x104d4400 Revision Id: 0x100005 No Modem Function Group found Default PCM: rates [0x5f0]: 32000 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D1 D2 D3 CLKSTOP EPSS Power: setting=D0, actual=D0 GPIO: io=2, o=0, i=0, unsolicited=1, wake=1 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x11: Stereo Device: name="ALC889 Analog", type="Audio", device=0 Converter: stream=8, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x11: Stereo Converter: stream=8, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Audio Output] wcaps 0x11: Stereo Converter: stream=8, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x05 [Audio Output] wcaps 0x11: Stereo Converter: stream=8, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x5f0]: 32000 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x07 [Audio Input] wcaps 0x10011b: Stereo Amp-In Control: name="Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Device: name="ALC889 Analog", type="Audio", device=0 Amp-In caps: ofs=0x10, nsteps=0x2e, stepsize=0x03, mute=1 Amp-In vals: [0x28 0x28] Converter: stream=4, channel=0 SDI-Select: 0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Connection: 1 0x24 Node 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x10, nsteps=0x2e, stepsize=0x03, mute=1 Amp-In vals: [0x90 0x90] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x10, nsteps=0x2e, stepsize=0x03, mute=1 Amp-In vals: [0x90 0x90] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Connection: 1 0x22 Node 0x0a [Audio Input] wcaps 0x100391: Stereo Digital Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x570]: 32000 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Unsolicited: tag=00, enabled=0 Connection: 1 0x1f Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Mic Playback Volume", index=1, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Mic Playback Switch", index=1, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Beep Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=5, ofs=0 Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=5, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x8f 0x8f] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 10 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x3e, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x3e 0x3e] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Surround Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Amp-Out caps: ofs=0x3e, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x3e 0x3e] Connection: 2 0x03 0x0b Node 0x0e [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Center Playback Volume", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="LFE Playback Volume", index=0, device=0 ControlAmp: chs=2, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Amp-Out caps: ofs=0x3e, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x3e 0x3e] Connection: 2 0x04 0x0b Node 0x0f [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Speaker Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x3e, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] Connection: 2 0x05 0x0b Node 0x10 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x5f0]: 32000 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x11 [Pin Complex] wcaps 0x400300: Mono Digital Pincap 0x00000010: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x10 Node 0x12 [Pin Complex] wcaps 0x400001: Stereo Control: name="Internal Mic Phantom Jack", index=0, device=0 Pincap 0x00000020: IN Pin Default 0x90a60120: [Fixed] Mic at Int N/A Conn = Digital, Color = Unknown DefAssociation = 0x2, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Speaker Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Speaker Phantom Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0001003c: IN OUT HP EAPD Detect EAPD 0x2: EAPD Pin Default 0x90170110: [Fixed] Speaker at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e 0x0f* 0x26 Node 0x15 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0001003c: IN OUT HP EAPD Detect EAPD 0x2: EAPD Pin Default 0x0421101f: [Jack] HP Out at Ext Right Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP Unsolicited: tag=01, enabled=1 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x16 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00000034: IN OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e* 0x0f 0x26 Node 0x17 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00000034: IN OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e 0x0f* 0x26 Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Surround Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Mic Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x04a1102e: [Jack] Mic at Ext Right Conn = 1/8, Color = Black DefAssociation = 0x2, Sequence = 0xe Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=02, enabled=1 Connection: 5 0x0c 0x0d 0x0e 0x0f 0x26* Node 0x19 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Center Playback Switch", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="LFE Playback Switch", index=0, device=0 ControlAmp: chs=2, dir=Out, idx=0, ofs=0 Control: name="Mic Boost Volume", index=1, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Mic Jack", index=1, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x04a11030: [Jack] Mic at Ext Right Conn = 1/8, Color = Black DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=03, enabled=1 Connection: 5 0x0c 0x0d 0x0e 0x0f 0x26* Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e 0x0f 0x26* Node 0x1b [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e 0x0f 0x26* Node 0x1c [Pin Complex] wcaps 0x400081: Stereo Pincap 0x00000024: IN Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Pincap 0x00000010: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x06 Node 0x1f [Pin Complex] wcaps 0x400280: Mono Digital Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=28 Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x24 [Audio Selector] wcaps 0x300101: Stereo Connection: 12 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b 0x12* Node 0x25 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x26 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Amp-Out caps: ofs=0x3e, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x3e 0x3e] Connection: 2 0x25 0x0b --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw---T+ 1 root audio 116, 0 Jul 3 19:35 /dev/snd/controlC0 crw-rw---T+ 1 root audio 116, 4 Jul 3 19:35 /dev/snd/hwC0D0 crw-rw---T+ 1 root audio 116, 24 Jul 3 19:36 /dev/snd/pcmC0D0c crw-rw---T+ 1 root audio 116, 16 Jul 3 19:36 /dev/snd/pcmC0D0p crw-rw---T+ 1 root audio 116, 1 Jul 3 19:35 /dev/snd/seq crw-rw---T+ 1 root audio 116, 33 Jul 3 19:35 /dev/snd/timer /dev/snd/by-path: total 0 drwxr-xr-x 2 root root 60 Jul 3 19:35 . drwxr-xr-x 3 root root 180 Jul 3 19:35 .. lrwxrwxrwx 1 root root 12 Jul 3 19:35 pci-0000:00:1b.0 -> ../controlC0 !!ALSA configuration files !!------------------------ !!System wide config file (/etc/asound.conf) pcm.pulse { type pulse } ctl.pulse { type pulse } pcm.!default { type pulse } ctl.!default { type pulse } !!Aplay/Arecord output !!-------------------- APLAY **** List of PLAYBACK Hardware Devices **** card 0: MID [HDA Intel MID], device 0: ALC889 Analog [ALC889 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: MID [HDA Intel MID], device 0: ALC889 Analog [ALC889 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [MID] Card hw:0 'MID'/'HDA Intel MID at 0xb6420000 irq 45' Mixer name : 'Realtek ALC889' Components : 'HDA:10ec0889,104d4400,00100005' Controls : 34 Simple ctrls : 18 Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 64 [100%] [0.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 62 [97%] [0.00dB] [on] Front Right: Playback 62 [97%] [0.00dB] [on] Simple mixer control 'Speaker',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 64 [100%] [2.00dB] [on] Front Right: Playback 64 [100%] [2.00dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume penum Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.00dB] Front Right: Playback 255 [100%] [0.00dB] Simple mixer control 'Surround',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 62 [97%] [0.00dB] [on] Front Right: Playback 62 [97%] [0.00dB] [on] Simple mixer control 'Center',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 62 [97%] [0.00dB] [on] Simple mixer control 'LFE',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 62 [97%] [0.00dB] [on] Simple mixer control 'Mic',0 Capabilities: pvolume pswitch cswitch cswitch-joined cswitch-exclusive penum Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Mono Limits: Playback 0 - 31 Mono: Capture [off] Front Left: Playback 15 [48%] [-12.00dB] [off] Front Right: Playback 15 [48%] [-12.00dB] [off] Simple mixer control 'Mic 1',0 Capabilities: cswitch cswitch-joined cswitch-exclusive penum Capture exclusive group: 0 Capture channels: Mono Mono: Capture [off] Simple mixer control 'Mic Boost',0 Capabilities: volume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] Simple mixer control 'Mic',1 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Mic Boost',1 Capabilities: volume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] Simple mixer control 'Beep',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch penum Capture channels: Front Left - Front Right Limits: Capture 0 - 46 Front Left: Capture 40 [87%] [24.00dB] [on] Front Right: Capture 40 [87%] [24.00dB] [on] Simple mixer control 'Auto-Mute Mode',0 Capabilities: enum Items: 'Disabled' 'Enabled' Item0: 'Enabled' Simple mixer control 'Channel Mode',0 Capabilities: enum Items: '2ch' '4ch' '6ch' Item0: '2ch' Simple mixer control 'Digital',0 Capabilities: cvolume penum Capture channels: Front Left - Front Right Limits: Capture 0 - 120 Front Left: Capture 0 [0%] [-30.00dB] Front Right: Capture 0 [0%] [-30.00dB] Simple mixer control 'Internal Mic',0 Capabilities: cswitch cswitch-joined cswitch-exclusive penum Capture exclusive group: 0 Capture channels: Mono Mono: Capture [on] !!Alsactl output !!-------------- --startcollapse-- state.MID { control.1 { iface MIXER name 'Channel Mode' value '2ch' comment { access 'read write' type ENUMERATED count 1 item.0 '2ch' item.1 '4ch' item.2 '6ch' } } control.2 { iface MIXER name 'Headphone Playback Volume' value.0 62 value.1 62 comment { access 'read write' type INTEGER count 2 range '0 - 64' dbmin -6200 dbmax 200 dbvalue.0 0 dbvalue.1 0 } } control.3 { iface MIXER name 'Headphone Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.4 { iface MIXER name 'Surround Playback Volume' value.0 62 value.1 62 comment { access 'read write' type INTEGER count 2 range '0 - 64' dbmin -6200 dbmax 200 dbvalue.0 0 dbvalue.1 0 } } control.5 { iface MIXER name 'Surround Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.6 { iface MIXER name 'Center Playback Volume' value 62 comment { access 'read write' type INTEGER count 1 range '0 - 64' dbmin -6200 dbmax 200 dbvalue.0 0 } } control.7 { iface MIXER name 'LFE Playback Volume' value 62 comment { access 'read write' type INTEGER count 1 range '0 - 64' dbmin -6200 dbmax 200 dbvalue.0 0 } } control.8 { iface MIXER name 'Center Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.9 { iface MIXER name 'LFE Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.10 { iface MIXER name 'Speaker Playback Volume' value.0 64 value.1 64 comment { access 'read write' type INTEGER count 2 range '0 - 64' dbmin -6200 dbmax 200 dbvalue.0 200 dbvalue.1 200 } } control.11 { iface MIXER name 'Speaker Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.12 { iface MIXER name 'Mic Playback Volume' value.0 15 value.1 15 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -1200 dbvalue.1 -1200 } } control.13 { iface MIXER name 'Mic Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.14 { iface MIXER name 'Mic Playback Volume' index 1 value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.15 { iface MIXER name 'Mic Playback Switch' index 1 value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.16 { iface MIXER name 'Auto-Mute Mode' value Enabled comment { access 'read write' type ENUMERATED count 1 item.0 Disabled item.1 Enabled } } control.17 { iface MIXER name 'Capture Source' value 'Internal Mic' comment { access 'read write' type ENUMERATED count 1 item.0 Mic item.1 'Mic 1' item.2 'Internal Mic' } } control.18 { iface MIXER name 'Capture Volume' value.0 40 value.1 40 comment { access 'read write' type INTEGER count 2 range '0 - 46' dbmin -1600 dbmax 3000 dbvalue.0 2400 dbvalue.1 2400 } } control.19 { iface MIXER name 'Capture Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.20 { iface MIXER name 'Mic Boost Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 3' dbmin 0 dbmax 3000 dbvalue.0 0 dbvalue.1 0 } } control.21 { iface MIXER name 'Mic Boost Volume' index 1 value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 3' dbmin 0 dbmax 3000 dbvalue.0 0 dbvalue.1 0 } } control.22 { iface MIXER name 'Master Playback Volume' value 64 comment { access 'read write' type INTEGER count 1 range '0 - 64' dbmin -6400 dbmax 0 dbvalue.0 0 } } control.23 { iface MIXER name 'Master Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.24 { iface CARD name 'Mic Jack' value false comment { access read type BOOLEAN count 1 } } control.25 { iface CARD name 'Mic Jack' index 1 value false comment { access read type BOOLEAN count 1 } } control.26 { iface CARD name 'Internal Mic Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.27 { iface CARD name 'Headphone Jack' value false comment { access read type BOOLEAN count 1 } } control.28 { iface CARD name 'Speaker Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.29 { iface MIXER name 'Beep Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.30 { iface MIXER name 'Beep Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.31 { iface PCM name 'Playback Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.32 { iface PCM name 'Capture Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.33 { iface MIXER name 'PCM Playback Volume' value.0 255 value.1 255 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '0000000100000008ffffec1400000014' dbmin -5100 dbmax 0 dbvalue.0 0 dbvalue.1 0 } } control.34 { iface MIXER name 'Digital Capture Volume' value.0 0 value.1 0 comment { access 'read write user' type INTEGER count 2 range '0 - 120' tlv '0000000100000008fffff44800000032' dbmin -3000 dbmax 3000 dbvalue.0 -3000 dbvalue.1 -3000 } } } --endcollapse-- !!All Loaded Modules !!------------------ Module parport_pc ppdev lp parport binfmt_misc fuse ext2 mbcache dm_crypt loop snd_hda_codec_realtek videobuf2_vmalloc videobuf2_memops videobuf2_core videodev media joydev snd_hda_intel snd_hda_codec snd_hwdep snd_pcm_oss snd_mixer_oss snd_pcm snd_page_alloc snd_seq_dummy snd_seq_oss snd_seq_midi snd_seq_midi_event snd_rawmidi snd_seq arc4 coretemp iwldvm kvm_intel mac80211 snd_seq_device kvm snd_timer crc32c_intel ghash_clmulni_intel snd iTCO_wdt iTCO_vendor_support aesni_intel iwlwifi i915 aes_x86_64 ablk_helper mxm_wmi cryptd xts cfg80211 sg lrw gf128mul sr_mod cdrom psmouse drm_kms_helper serio_raw sony_laptop evdev drm rfkill pcspkr video wmi i2c_i801 i2c_algo_bit battery button i2c_core ehci_pci ehci_hcd e1000e intel_ips soundcore acpi_cpufreq mperf ac lpc_ich usbcore mfd_core mei sdhci_pci sdhci ptp mmc_core pps_core usb_common processor btrfs crc32c libcrc32c xor zlib_deflate raid6_pq dm_mod raid0 md_mod sd_mod crc_t10dif ahci libahci microcode thermal thermal_sys libata scsi_mod !!Sysfs Files !!----------- /sys/class/sound/hwC0D0/init_pin_configs: 0x11 0x411111f0 0x12 0x90a60120 0x14 0x90170110 0x15 0x0421101f 0x16 0x411111f0 0x17 0x411111f0 0x18 0x04a1102e 0x19 0x04a11030 0x1a 0x411111f0 0x1b 0x411111f0 0x1c 0x411111f0 0x1d 0x411111f0 0x1e 0x411111f0 0x1f 0x411111f0 /sys/class/sound/hwC0D0/driver_pin_configs: /sys/class/sound/hwC0D0/user_pin_configs: /sys/class/sound/hwC0D0/init_verbs: /sys/class/sound/hwC0D0/hints: !!ALSA/HDA dmesg !!-------------- [ 5.121872] pipe_off wait timed out [ 5.121921] Modules linked in: uvcvideo videobuf2_vmalloc videobuf2_memops videobuf2_core videodev media joydev snd_hda_intel(O+) snd_hda_codec(O) snd_hwdep(O) snd_pcm_oss(O) snd_mixer_oss(O) snd_pcm(O) snd_page_alloc(O) snd_seq_dummy(O) snd_seq_oss(O) snd_seq_midi(O) snd_seq_midi_event(O) snd_rawmidi(O) snd_seq(O) arc4 coretemp iwldvm kvm_intel mac80211 snd_seq_device(O) kvm snd_timer(O) crc32c_intel ghash_clmulni_intel snd(O) iTCO_wdt iTCO_vendor_support aesni_intel iwlwifi i915(+) aes_x86_64 ablk_helper mxm_wmi cryptd xts cfg80211 sg lrw gf128mul sr_mod cdrom psmouse drm_kms_helper serio_raw sony_laptop evdev drm rfkill pcspkr video wmi i2c_i801 i2c_algo_bit battery button i2c_core ehci_pci ehci_hcd e1000e intel_ips soundcore acpi_cpufreq mperf ac lpc_ich usbcore mfd_core mei sdhci_pci sdhci ptp mmc_core pps_core usb_common processor btrfs crc32c libcrc32c xor zlib_deflate raid6_pq dm_mod raid0 md_mod sd_mod crc_t10dif ahci libahci microcode thermal thermal_sys libata scsi_mod [ 5.121941] Pid: 635, comm: modprobe Tainted: G O 3.9-0.bpo.1-amd64 #1 Debian 3.9.6-1~bpo70+1 -- [ 5.835404] [drm] Initialized i915 1.6.0 20080730 for 0000:00:02.0 on minor 0 [ 5.835752] snd_hda_intel 0000:00:1b.0: irq 45 for MSI/MSI-X [ 5.864320] hda_codec: ALC889: SKU not ready 0x411111f0 [ 5.864675] ALSA hda_auto_parser.c:393 autoconfig: line_outs=1 (0x14/0x0/0x0/0x0/0x0) type:speaker [ 5.864681] ALSA hda_auto_parser.c:397 speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 5.864686] ALSA hda_auto_parser.c:401 hp_outs=1 (0x15/0x0/0x0/0x0/0x0) [ 5.864689] ALSA hda_auto_parser.c:402 mono: mono_out=0x0 [ 5.864693] ALSA hda_auto_parser.c:406 inputs: [ 5.864697] ALSA hda_auto_parser.c:410 Mic=0x18 [ 5.864701] ALSA hda_auto_parser.c:410 Mic=0x19 [ 5.864704] ALSA hda_auto_parser.c:410 Internal Mic=0x12 [ 5.864709] ALSA patch_realtek.c:486 realtek: No valid SSID, checking pincfg 0x411111f0 for NID 0x1d [ 5.864713] ALSA patch_realtek.c:569 realtek: Enable default setup for auto mode as fallback [ 5.868644] input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:1b.0/input/input9 [ 5.874177] input: HDA Intel MID Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card0/input10 [ 5.874232] input: HDA Intel MID Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input11 [ 5.874284] input: HDA Intel MID Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input12 [ 6.760494] btrfs: use ssd allocation scheme -- [ 26.166424] pipe_off wait timed out [ 26.166426] Modules linked in: parport_pc ppdev lp parport binfmt_misc fuse ext2 mbcache dm_crypt loop snd_hda_codec_realtek(O) uvcvideo videobuf2_vmalloc videobuf2_memops videobuf2_core videodev media joydev snd_hda_intel(O) snd_hda_codec(O) snd_hwdep(O) snd_pcm_oss(O) snd_mixer_oss(O) snd_pcm(O) snd_page_alloc(O) snd_seq_dummy(O) snd_seq_oss(O) snd_seq_midi(O) snd_seq_midi_event(O) snd_rawmidi(O) snd_seq(O) arc4 coretemp iwldvm kvm_intel mac80211 snd_seq_device(O) kvm snd_timer(O) crc32c_intel ghash_clmulni_intel snd(O) iTCO_wdt iTCO_vendor_support aesni_intel iwlwifi i915 aes_x86_64 ablk_helper mxm_wmi cryptd xts cfg80211 sg lrw gf128mul sr_mod cdrom psmouse drm_kms_helper serio_raw sony_laptop evdev drm rfkill pcspkr video wmi i2c_i801 i2c_algo_bit battery button i2c_core ehci_pci ehci_hcd e1000e intel_ips soundcore acpi_cpufreq mperf ac lpc_ich usbcore mfd_core mei sdhci_pci sdhci ptp mmc_core pps_core usb_common processor btrfs crc32c libcrc32c xor zlib_deflate raid6_pq dm_mod raid0 md_mod sd_mod crc_t10dif ahci libahci microcode thermal thermal_sys libata scsi_mod [ 26.166527] Pid: 4157, comm: Xorg Tainted: G W O 3.9-0.bpo.1-amd64 #1 Debian 3.9.6-1~bpo70+1

4 38

[alsa-devel] [PATCH 1/2] ASoC: 88pm60x: Don't use control data for i2c
by Mark Brown 19 Sep '13

19 Sep '13

From: Mark Brown <broonie(a)linaro.org> In preparation for using the regmap directly in the CODEC driver replace references to the I2C client using control_data with references to the driver private data. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/88pm860x-codec.c | 17 +++++++++-------- 1 file changed, 9 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 259d1ac..ca5f9f0 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1169,6 +1169,7 @@ static int pm860x_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai, static int pm860x_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec); int data; switch (level) { @@ -1182,17 +1183,17 @@ static int pm860x_set_bias_level(struct snd_soc_codec *codec, if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* Enable Audio PLL & Audio section */ data = AUDIO_PLL | AUDIO_SECTION_ON; - pm860x_reg_write(codec->control_data, REG_MISC2, data); + pm860x_reg_write(pm860x->i2c, REG_MISC2, data); udelay(300); data = AUDIO_PLL | AUDIO_SECTION_RESET | AUDIO_SECTION_ON; - pm860x_reg_write(codec->control_data, REG_MISC2, data); + pm860x_reg_write(pm860x->i2c, REG_MISC2, data); } break; case SND_SOC_BIAS_OFF: data = AUDIO_PLL | AUDIO_SECTION_RESET | AUDIO_SECTION_ON; - pm860x_set_bits(codec->control_data, REG_MISC2, data, 0); + pm860x_set_bits(pm860x->i2c, REG_MISC2, data, 0); break; } codec->dapm.bias_level = level; @@ -1322,17 +1323,17 @@ int pm860x_hs_jack_detect(struct snd_soc_codec *codec, pm860x->det.lo_shrt = lo_shrt; if (det & SND_JACK_HEADPHONE) - pm860x_set_bits(codec->control_data, REG_HS_DET, + pm860x_set_bits(pm860x->i2c, REG_HS_DET, EN_HS_DET, EN_HS_DET); /* headset short detect */ if (hs_shrt) { data = CLR_SHORT_HS2 | CLR_SHORT_HS1; - pm860x_set_bits(codec->control_data, REG_SHORTS, data, data); + pm860x_set_bits(pm860x->i2c, REG_SHORTS, data, data); } /* Lineout short detect */ if (lo_shrt) { data = CLR_SHORT_LO2 | CLR_SHORT_LO1; - pm860x_set_bits(codec->control_data, REG_SHORTS, data, data); + pm860x_set_bits(pm860x->i2c, REG_SHORTS, data, data); } /* sync status */ @@ -1350,7 +1351,7 @@ int pm860x_mic_jack_detect(struct snd_soc_codec *codec, pm860x->det.mic_det = det; if (det & SND_JACK_MICROPHONE) - pm860x_set_bits(codec->control_data, REG_MIC_DET, + pm860x_set_bits(pm860x->i2c, REG_MIC_DET, MICDET_MASK, MICDET_MASK); /* sync status */ @@ -1380,7 +1381,7 @@ static int pm860x_probe(struct snd_soc_codec *codec) pm860x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - ret = pm860x_bulk_read(codec->control_data, REG_CACHE_BASE, + ret = pm860x_bulk_read(pm860x->i2c, REG_CACHE_BASE, REG_CACHE_SIZE, codec->reg_cache); if (ret < 0) { dev_err(codec->dev, "Failed to fill register cache: %d\n", -- 1.8.4.rc3

1 1

[alsa-devel] [RFC PATCH] ES938 support for ES18xx driver
by Ondrej Zary 19 Sep '13

19 Sep '13

Hello, this is an attempt to support ES938 3-D Audio Effects Processor found on some ES18xx (and possibly other) sound cards, doing bass/treble and 3D control. ES938 is controlled by MIDI SysEx commands sent through card's MPU401 port. The following patch works but has a problem: the midi port cannot be used by userspace applications. Opening and closing the rawmidi device on each control change would probably work (as long as the port is not used by an application) but that just does not seem right. Is there a way to cleanly fix this? (There are two problems: 3D level does not have any effect and alsamixer does not display the TLV dB values - but don't know why). --- sound/isa/Kconfig | 4 + sound/isa/Makefile | 2 + sound/isa/es18xx.c | 12 +++- sound/isa/es938.c | 181 ++++++++++++++++++++++++++++++++++++++++++++++++++++ sound/isa/es938.h | 24 +++++++ 5 files changed, 222 insertions(+), 1 deletions(-) create mode 100644 sound/isa/es938.c create mode 100644 sound/isa/es938.h diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index affa134..a8bf3e3 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -17,6 +17,9 @@ config SND_SB16_DSP select SND_PCM select SND_SB_COMMON +config SND_ES938 + tristate + menuconfig SND_ISA bool "ISA sound devices" depends on ISA && ISA_DMA_API @@ -183,6 +186,7 @@ config SND_ES18XX select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM + select SND_ES938 help Say Y here to include support for ESS AudioDrive ES18xx chips. diff --git a/sound/isa/Makefile b/sound/isa/Makefile index 9a15f14..d59e0bf 100644 --- a/sound/isa/Makefile +++ b/sound/isa/Makefile @@ -8,6 +8,7 @@ snd-als100-objs := als100.o snd-azt2320-objs := azt2320.o snd-cmi8328-objs := cmi8328.o snd-cmi8330-objs := cmi8330.o +snd-es938-objs := es938.o snd-es18xx-objs := es18xx.o snd-opl3sa2-objs := opl3sa2.o snd-sc6000-objs := sc6000.o @@ -19,6 +20,7 @@ obj-$(CONFIG_SND_ALS100) += snd-als100.o obj-$(CONFIG_SND_AZT2320) += snd-azt2320.o obj-$(CONFIG_SND_CMI8328) += snd-cmi8328.o obj-$(CONFIG_SND_CMI8330) += snd-cmi8330.o +obj-$(CONFIG_SND_ES938) += snd-es938.o obj-$(CONFIG_SND_ES18XX) += snd-es18xx.o obj-$(CONFIG_SND_OPL3SA2) += snd-opl3sa2.o obj-$(CONFIG_SND_SC6000) += snd-sc6000.o diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 12978b8..c502aa0 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -96,6 +96,7 @@ #define SNDRV_LEGACY_FIND_FREE_IRQ #define SNDRV_LEGACY_FIND_FREE_DMA #include <sound/initval.h> +#include "es938.h" #define PFX "es18xx: " @@ -122,6 +123,7 @@ struct snd_es18xx { struct snd_pcm_substream *playback_b_substream; struct snd_rawmidi *rmidi; + struct snd_es938 es938; struct snd_kcontrol *hw_volume; struct snd_kcontrol *hw_switch; @@ -2166,7 +2168,15 @@ static int snd_audiodrive_probe(struct snd_card *card, int dev) return err; } - return snd_card_register(card); + err = snd_card_register(card); + if (err < 0) + return err; + + if (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT) + if (snd_es938_init(&chip->es938, card, 0, 0) == 0) + printk("es938 found!\n"); + + return 0; } static int snd_es18xx_isa_match(struct device *pdev, unsigned int dev) diff --git a/sound/isa/es938.c b/sound/isa/es938.c new file mode 100644 index 0000000..cfe8ae7 --- /dev/null +++ b/sound/isa/es938.c @@ -0,0 +1,181 @@ +/* + * Driver for ESS ES938 3-D Audio Effects Processor + * Copyright (c) 2013 Ondrej Zary + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +//#include <linux/init.h> +#include <linux/module.h> +#include <sound/asoundef.h> +#include <sound/control.h> +#include <sound/core.h> +#include <sound/rawmidi.h> +#include "es938.h" + +#define PFX "es938: " + +MODULE_AUTHOR("Ondrej Zary"); +MODULE_DESCRIPTION("ESS ES938"); +MODULE_LICENSE("GPL"); + +static int snd_es938_read_reg(struct snd_es938 *chip, u8 reg, u8 *out) +{ + u8 buf[8]; + int i = 0, res; + u8 sysex[] = { MIDI_CMD_COMMON_SYSEX, ES938_ID, ES938_CMD_REG_R, reg, + MIDI_CMD_COMMON_SYSEX_END }; + + snd_rawmidi_kernel_write(chip->rfile.output, sysex, sizeof(sysex)); + + memset(buf, 0, sizeof(buf)); + while (i < sizeof(buf)) { + res = snd_rawmidi_kernel_read(chip->rfile.input, buf + i, sizeof(buf) - i); + if (res > 0) + i+= res; + } + + /* check reply */ + if (memcmp(buf, sysex, 6) || buf[7] != MIDI_CMD_COMMON_SYSEX_END) + return -1; + + if (out) + *out = buf[6]; + + return 0; +} + +static void snd_es938_write_reg(struct snd_es938 *chip, u8 reg, u8 val) +{ + u8 sysex[] = { MIDI_CMD_COMMON_SYSEX, ES938_ID, ES938_CMD_REG_W, reg, + val, MIDI_CMD_COMMON_SYSEX_END }; + + snd_rawmidi_kernel_write(chip->rfile.output, sysex, sizeof(sysex)); + chip->regs[reg] = val; +} + +static void snd_es938_write_reg_mask(struct snd_es938 *chip, u8 reg, u8 val, u8 mask) +{ + u8 oldval = chip->regs[reg]; + oldval &= ~mask; + snd_es938_write_reg(chip, reg, oldval | (val & mask)); +} + +#define ES938_MIXER(xname, xindex, reg, shift, mask) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ + .info = snd_es938_info_mixer, \ + .get = snd_es938_get_mixer, .put = snd_es938_put_mixer, \ + .private_value = reg | (shift << 8) | (mask << 16) } + +#define ES938_MIXER_TLV(xname, xindex, reg, shift, mask, xtlv) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ + .info = snd_es938_info_mixer, \ + .get = snd_es938_get_mixer, .put = snd_es938_put_mixer, \ + .private_value = reg | (shift << 8) | (mask << 16), \ + .tlv = { .p = (xtlv) } } + +static const DECLARE_TLV_DB_SCALE(db_scale_tone, -900, 300, 0); + +static struct snd_kcontrol_new snd_es938_controls[] = { +ES938_MIXER_TLV("Tone Control - Bass", 0, ES938_REG_TONE, 0, 7, db_scale_tone), +ES938_MIXER_TLV("Tone Control - Treble", 0, ES938_REG_TONE, 4, 7, db_scale_tone), +ES938_MIXER("3D Control - Level", 0, ES938_REG_SPATIAL, 1, 63), +ES938_MIXER("3D Control - Switch", 0, ES938_REG_SPATIAL_EN, 0, 1), +}; + +int snd_es938_init(struct snd_es938 *chip, struct snd_card *card, int device, + int subdevice) +{ + int err, i; + + if ((err = snd_rawmidi_kernel_open(card, device, subdevice, + SNDRV_RAWMIDI_LFLG_OPEN | SNDRV_RAWMIDI_LFLG_APPEND, + &chip->rfile)) < 0) { + snd_printk(KERN_WARNING PFX "unable to open MIDI device\n"); + return err; + } + + /* try to read a register to detect chip presence */ + if (snd_es938_read_reg(chip, ES938_REG_MISC, NULL) < 0) + return -ENODEV; + + /* write default values (there's no reset) */ + snd_es938_write_reg(chip, ES938_REG_MISC, 0x49); + snd_es938_write_reg(chip, ES938_REG_TONE, 0x33); + snd_es938_write_reg(chip, ES938_REG_SPATIAL, 0x00); + /* datasheet specifies invalid value 0x00 as default */ + snd_es938_write_reg(chip, ES938_REG_SPATIAL_EN, 0x02); + snd_es938_write_reg(chip, ES938_REG_POWER, 0x0e); + + strlcat(card->mixername, " + ES938", sizeof(card->mixername)); + for (i = 0; i < ARRAY_SIZE(snd_es938_controls); i++) + if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_es938_controls[i], chip))) < 0) + return err; + + return 0; +} +EXPORT_SYMBOL(snd_es938_init); + +void snd_es938_release(struct snd_es938 *chip) +{ + snd_rawmidi_kernel_release(&chip->rfile); +} +EXPORT_SYMBOL(snd_es938_release); + + +int snd_es938_info_mixer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + int mask = (kcontrol->private_value >> 16) & 0xff; + + uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = mask; + return 0; +} +EXPORT_SYMBOL(snd_es938_info_mixer); + +int snd_es938_get_mixer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct snd_es938 *chip = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int shift = (kcontrol->private_value >> 8) & 0xff; + int mask = (kcontrol->private_value >> 16) & 0xff; + u8 val = chip->regs[reg]; + + ucontrol->value.integer.value[0] = (val >> shift) & mask; + return 0; +} +EXPORT_SYMBOL(snd_es938_get_mixer); + +int snd_es938_put_mixer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct snd_es938 *chip = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int shift = (kcontrol->private_value >> 8) & 0xff; + int mask = (kcontrol->private_value >> 16) & 0xff; + u8 val = ucontrol->value.integer.value[0] & mask; + u8 oldval = chip->regs[reg]; + + mask <<= shift; + val <<= shift; + + snd_es938_write_reg_mask(chip, reg, val, mask); + return chip->regs[reg] != oldval; +} +EXPORT_SYMBOL(snd_es938_put_mixer); diff --git a/sound/isa/es938.h b/sound/isa/es938.h new file mode 100644 index 0000000..1f8595f --- /dev/null +++ b/sound/isa/es938.h @@ -0,0 +1,24 @@ +#include <sound/tlv.h> + +#define ES938_ID 0x00, 0x00, 0x7b + +#define ES938_CMD_REG_R 0x7e +#define ES938_CMD_REG_W 0x7f + +#define ES938_REG_MISC 0 +#define ES938_REG_TONE 1 +#define ES938_REG_SPATIAL 5 +#define ES938_REG_SPATIAL_EN 6 +#define ES938_REG_POWER 7 + +struct snd_es938 { + u8 regs[8]; + struct snd_rawmidi_file rfile; +}; + +int snd_es938_init(struct snd_es938 *chip, struct snd_card *card, int device, + int subdevice); +void snd_es938_release(struct snd_es938 *chip); +int snd_es938_info_mixer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); +int snd_es938_get_mixer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_es938_put_mixer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); -- Ondrej Zary

2 8

[alsa-devel] [PATCH] Fix access of freed memory in namehints
by David Henningsson 19 Sep '13

19 Sep '13

Sometimes a hook manipulates the config tree, which makes currently running iterators point to freed memory. As a workaround, make two copies, one for the iterators and another for the hooks. BugLink: https://bugs.launchpad.net/bugs/1008600 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- src/control/namehint.c | 22 +++++++++++++--------- 1 file changed, 13 insertions(+), 9 deletions(-) diff --git a/src/control/namehint.c b/src/control/namehint.c index 8d5e925..28975a4 100644 --- a/src/control/namehint.c +++ b/src/control/namehint.c @@ -406,7 +406,7 @@ static const next_devices_t next_devices[] = { }; #endif -static int add_card(snd_config_t *config, struct hint_list *list, int card) +static int add_card(snd_config_t *config, snd_config_t *rw_config, struct hint_list *list, int card) { int err, ok; snd_config_t *conf, *n; @@ -449,7 +449,7 @@ static int add_card(snd_config_t *config, struct hint_list *list, int card) ok = 0; for (device = 0; err >= 0 && device <= max_device; device++) { list->device = device; - err = try_config(config, list, list->siface, str); + err = try_config(rw_config, list, list->siface, str); if (err < 0) break; ok++; @@ -464,7 +464,7 @@ static int add_card(snd_config_t *config, struct hint_list *list, int card) if (err < 0) { list->card = card; list->device = -1; - err = try_config(config, list, list->siface, str); + err = try_config(rw_config, list, list->siface, str); } if (err == -ENOMEM) goto __error; @@ -493,7 +493,8 @@ static int get_card_name(struct hint_list *list, int card) return 0; } -static int add_software_devices(snd_config_t *config, struct hint_list *list) +static int add_software_devices(snd_config_t *config, snd_config_t *rw_config, + struct hint_list *list) { int err; snd_config_t *conf, *n; @@ -509,7 +510,7 @@ static int add_software_devices(snd_config_t *config, struct hint_list *list) continue; list->card = -1; list->device = -1; - err = try_config(config, list, list->siface, str); + err = try_config(rw_config, list, list->siface, str); if (err == -ENOMEM) return -ENOMEM; } @@ -547,7 +548,7 @@ int snd_device_name_hint(int card, const char *iface, void ***hints) struct hint_list list; char ehints[24]; const char *str; - snd_config_t *conf, *local_config = NULL; + snd_config_t *conf, *local_config = NULL, *local_config_rw = NULL; snd_config_update_t *local_config_update = NULL; snd_config_iterator_t i, next; int err; @@ -557,6 +558,7 @@ int snd_device_name_hint(int card, const char *iface, void ***hints) err = snd_config_update_r(&local_config, &local_config_update, NULL); if (err < 0) return err; + err = snd_config_copy(&local_config_rw, local_config); list.list = NULL; list.count = list.allocated = 0; list.siface = iface; @@ -586,9 +588,9 @@ int snd_device_name_hint(int card, const char *iface, void ***hints) if (card >= 0) { err = get_card_name(&list, card); if (err >= 0) - err = add_card(local_config, &list, card); + err = add_card(local_config, local_config_rw, &list, card); } else { - add_software_devices(local_config, &list); + add_software_devices(local_config, local_config_rw, &list); err = snd_card_next(&card); if (err < 0) goto __error; @@ -596,7 +598,7 @@ int snd_device_name_hint(int card, const char *iface, void ***hints) err = get_card_name(&list, card); if (err < 0) goto __error; - err = add_card(local_config, &list, card); + err = add_card(local_config, local_config_rw, &list, card); if (err < 0) goto __error; err = snd_card_next(&card); @@ -630,6 +632,8 @@ int snd_device_name_hint(int card, const char *iface, void ***hints) if (list.cardname) free(list.cardname); } + if (local_config_rw) + snd_config_delete(local_config_rw); if (local_config) snd_config_delete(local_config); if (local_config_update) -- 1.8.3.2

2 1

[alsa-devel] question about webcam audio capture
by vichy 19 Sep '13

19 Sep '13

hi all: I try to do webcam audio capture on my arm platform. when I execute audio capture I got below message "retire_capture_urb: 108 callbacks suppressed" my questions: 1. where I can get the message "retire_capture_urb: 108 callbacks suppressed" show in sound/usb driver? I try to grep -rwn '"retire_capture_urb.*' sound/usb/*, but get nothing. 2. from usb mon log, it seems HW keep Active status high even until ehci_io_watchdog happen, because the ep status is -EXDEV. if my observation is correct, is it possible to modify usb sound driver to purposely send fix number urbs. Because from the usb mon log, the first 8 urb will get time out already. thanks for your help in advance,

2 7

[alsa-devel] [PATCH 1/1] ASoC: mmp-sspa: Use devm_snd_soc_register_component
by Sachin Kamat 19 Sep '13

19 Sep '13

devm_snd_soc_register_component makes code simpler. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/pxa/mmp-sspa.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c index 41752a5..5bf5f1f 100644 --- a/sound/soc/pxa/mmp-sspa.c +++ b/sound/soc/pxa/mmp-sspa.c @@ -455,8 +455,8 @@ static int asoc_mmp_sspa_probe(struct platform_device *pdev) priv->dai_fmt = (unsigned int) -1; platform_set_drvdata(pdev, priv); - return snd_soc_register_component(&pdev->dev, &mmp_sspa_component, - &mmp_sspa_dai, 1); + return devm_snd_soc_register_component(&pdev->dev, &mmp_sspa_component, + &mmp_sspa_dai, 1); } static int asoc_mmp_sspa_remove(struct platform_device *pdev) @@ -466,7 +466,6 @@ static int asoc_mmp_sspa_remove(struct platform_device *pdev) clk_disable(priv->audio_clk); clk_put(priv->audio_clk); clk_put(priv->sysclk); - snd_soc_unregister_component(&pdev->dev); return 0; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/1] ASoC: omap-mcpdm: Use devm_snd_soc_register_component
by Sachin Kamat 19 Sep '13

19 Sep '13

devm_snd_soc_register_component makes code simpler. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- Compile tested. --- sound/soc/omap/omap-mcpdm.c | 12 +++--------- 1 file changed, 3 insertions(+), 9 deletions(-) diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 90d2a7c..cd9ee16 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -490,14 +490,9 @@ static int asoc_mcpdm_probe(struct platform_device *pdev) mcpdm->dev = &pdev->dev; - return snd_soc_register_component(&pdev->dev, &omap_mcpdm_component, - &omap_mcpdm_dai, 1); -} - -static int asoc_mcpdm_remove(struct platform_device *pdev) -{ - snd_soc_unregister_component(&pdev->dev); - return 0; + return devm_snd_soc_register_component(&pdev->dev, + &omap_mcpdm_component, + &omap_mcpdm_dai, 1); } static const struct of_device_id omap_mcpdm_of_match[] = { @@ -514,7 +509,6 @@ static struct platform_driver asoc_mcpdm_driver = { }, .probe = asoc_mcpdm_probe, - .remove = asoc_mcpdm_remove, }; module_platform_driver(asoc_mcpdm_driver); -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/1] ASoC: mxs-saif: Use devm_snd_soc_register_component
by Sachin Kamat 19 Sep '13

19 Sep '13

devm_snd_soc_register_component makes code simpler. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/mxs/mxs-saif.c | 12 +++--------- 1 file changed, 3 insertions(+), 9 deletions(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index b56b8a0..14152f6 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -768,8 +768,8 @@ static int mxs_saif_probe(struct platform_device *pdev) dev_warn(&pdev->dev, "failed to init clocks\n"); } - ret = snd_soc_register_component(&pdev->dev, &mxs_saif_component, - &mxs_saif_dai, 1); + ret = devm_snd_soc_register_component(&pdev->dev, &mxs_saif_component, + &mxs_saif_dai, 1); if (ret) { dev_err(&pdev->dev, "register DAI failed\n"); return ret; @@ -778,21 +778,15 @@ static int mxs_saif_probe(struct platform_device *pdev) ret = mxs_pcm_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "register PCM failed: %d\n", ret); - goto failed_pdev_alloc; + return ret; } return 0; - -failed_pdev_alloc: - snd_soc_unregister_component(&pdev->dev); - - return ret; } static int mxs_saif_remove(struct platform_device *pdev) { mxs_pcm_platform_unregister(&pdev->dev); - snd_soc_unregister_component(&pdev->dev); return 0; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/1] ASoC: fsl_spdif: Use devm_snd_soc_register_component
by Sachin Kamat 19 Sep '13

19 Sep '13

devm_snd_soc_register_component makes code simpler. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/fsl/fsl_spdif.c | 14 +++----------- 1 file changed, 3 insertions(+), 11 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 3920c3e..44378e6 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1172,23 +1172,16 @@ static int fsl_spdif_probe(struct platform_device *pdev) /* Register with ASoC */ dev_set_drvdata(&pdev->dev, spdif_priv); - ret = snd_soc_register_component(&pdev->dev, &fsl_spdif_component, - &spdif_priv->cpu_dai_drv, 1); + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_spdif_component, + &spdif_priv->cpu_dai_drv, 1); if (ret) { dev_err(&pdev->dev, "failed to register DAI: %d\n", ret); return ret; } ret = imx_pcm_dma_init(pdev); - if (ret) { + if (ret) dev_err(&pdev->dev, "imx_pcm_dma_init failed: %d\n", ret); - goto error_component; - } - - return ret; - -error_component: - snd_soc_unregister_component(&pdev->dev); return ret; } @@ -1196,7 +1189,6 @@ error_component: static int fsl_spdif_remove(struct platform_device *pdev) { imx_pcm_dma_exit(pdev); - snd_soc_unregister_component(&pdev->dev); return 0; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/1] ASoC: omap-twl4030: Use devm_snd_soc_register_card
by Sachin Kamat 19 Sep '13

19 Sep '13

devm_snd_soc_register_card makes code simpler. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/omap/omap-twl4030.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c index 2a9324f..6a8d6b5 100644 --- a/sound/soc/omap/omap-twl4030.c +++ b/sound/soc/omap/omap-twl4030.c @@ -338,9 +338,9 @@ static int omap_twl4030_probe(struct platform_device *pdev) } snd_soc_card_set_drvdata(card, priv); - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + dev_err(&pdev->dev, "devm_snd_soc_register_card() failed: %d\n", ret); return ret; } @@ -357,7 +357,6 @@ static int omap_twl4030_remove(struct platform_device *pdev) snd_soc_jack_free_gpios(&priv->hs_jack, ARRAY_SIZE(hs_jack_gpios), hs_jack_gpios); - snd_soc_unregister_card(card); return 0; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/1] ASoC: mfld: Use devm_snd_soc_register_card
by Sachin Kamat 19 Sep '13

19 Sep '13

devm_snd_soc_register_card makes code simpler. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/mid-x86/mfld_machine.c | 10 +--------- 1 file changed, 1 insertion(+), 9 deletions(-) diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index ee36384..d3d4c32 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -400,7 +400,7 @@ static int snd_mfld_mc_probe(struct platform_device *pdev) } /* register the soc card */ snd_soc_card_mfld.dev = &pdev->dev; - ret_val = snd_soc_register_card(&snd_soc_card_mfld); + ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_mfld); if (ret_val) { pr_debug("snd_soc_register_card failed %d\n", ret_val); return ret_val; @@ -410,20 +410,12 @@ static int snd_mfld_mc_probe(struct platform_device *pdev) return 0; } -static int snd_mfld_mc_remove(struct platform_device *pdev) -{ - pr_debug("snd_mfld_mc_remove called\n"); - snd_soc_unregister_card(&snd_soc_card_mfld); - return 0; -} - static struct platform_driver snd_mfld_mc_driver = { .driver = { .owner = THIS_MODULE, .name = "msic_audio", }, .probe = snd_mfld_mc_probe, - .remove = snd_mfld_mc_remove, }; module_platform_driver(snd_mfld_mc_driver); -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/3] ASoC: imx-sgtl5000: Use devm_snd_soc_register_card
by Sachin Kamat 19 Sep '13

19 Sep '13

devm_snd_soc_register_card makes code simpler. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- This series compile tested. --- sound/soc/fsl/imx-sgtl5000.c | 12 +----------- 1 file changed, 1 insertion(+), 11 deletions(-) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 46c5b4f..78f86d8 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -159,7 +159,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) data->card.dapm_widgets = imx_sgtl5000_dapm_widgets; data->card.num_dapm_widgets = ARRAY_SIZE(imx_sgtl5000_dapm_widgets); - ret = snd_soc_register_card(&data->card); + ret = devm_snd_soc_register_card(&pdev->dev, &data->card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); goto fail; @@ -180,15 +180,6 @@ fail: return ret; } -static int imx_sgtl5000_remove(struct platform_device *pdev) -{ - struct imx_sgtl5000_data *data = platform_get_drvdata(pdev); - - snd_soc_unregister_card(&data->card); - - return 0; -} - static const struct of_device_id imx_sgtl5000_dt_ids[] = { { .compatible = "fsl,imx-audio-sgtl5000", }, { /* sentinel */ } @@ -202,7 +193,6 @@ static struct platform_driver imx_sgtl5000_driver = { .of_match_table = imx_sgtl5000_dt_ids, }, .probe = imx_sgtl5000_probe, - .remove = imx_sgtl5000_remove, }; module_platform_driver(imx_sgtl5000_driver); -- 1.7.9.5

2 3

Re: [alsa-devel] [PATCH] ASoC: rt5640: Omit ACPI match table only if !ACPI
by Mark Brown 19 Sep '13

19 Sep '13

On Thu, Sep 19, 2013 at 01:21:04PM +0200, Thierry Reding wrote: > Or perhaps we can annotate the table so that the compiler will stop > complaining but remove the table nonetheless if it isn't used. Similarly > to what we can do with > > if (IS_ENABLED(CONFIG_FOO)) { > ... > } > > constructs. I can't really think of a way to do that, though. Perhaps > gcc will throw away the table anyway, so we'll just need a way to make > it shut up? There's an annotation which can stop the unreferenced variable warning and then modern compilers are able to eliminate unreferenced static symbols. Not sure if it's actually worth the effort of doing all this though, but it is nice to be able to avoid needing multiple builds.

1 0

[alsa-devel] [PATCH] ASoC: fsl: imx-sgtl5000: fix error path if codec driver not found
by Andrew Gabbasov 19 Sep '13

19 Sep '13

Handling of this error condition should make some cleanup on function exit instead of immediate return. Signed-off-by: Andrew Gabbasov <andrew_gabbasov(a)mentor.com> --- sound/soc/fsl/imx-sgtl5000.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 46c5b4f..7ea5658 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -119,7 +119,8 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) codec_dev = of_find_i2c_device_by_node(codec_np); if (!codec_dev) { dev_err(&pdev->dev, "failed to find codec platform device\n"); - return -EPROBE_DEFER; + ret = -EPROBE_DEFER; + goto fail; } data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: fsl: imx-wm8962: defer probing if ssi/codec not found
by Andrew Gabbasov 19 Sep '13

19 Sep '13

If the ssi or codec driver is not yet loaded by the time of imx-wm8962 driver probing, this probe should be deferred until the dependent drivers loading. Return -EPROBE_DEFER in this case. Signed-off-by: Andrew Gabbasov <andrew_gabbasov(a)mentor.com> --- sound/soc/fsl/imx-wm8962.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 722afe6..9e60aa8 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -211,13 +211,13 @@ static int imx_wm8962_probe(struct platform_device *pdev) ssi_pdev = of_find_device_by_node(ssi_np); if (!ssi_pdev) { dev_err(&pdev->dev, "failed to find SSI platform device\n"); - ret = -EINVAL; + ret = -EPROBE_DEFER; goto fail; } codec_dev = of_find_i2c_device_by_node(codec_np); if (!codec_dev || !codec_dev->driver) { dev_err(&pdev->dev, "failed to find codec platform device\n"); - ret = -EINVAL; + ret = -EPROBE_DEFER; goto fail; } -- 1.7.10.4

1 0

[alsa-devel] [PATCH 1/3] mfd: mc13xxx: Don't require lock for simple register I/O
by Mark Brown 19 Sep '13

19 Sep '13

From: Mark Brown <broonie(a)linaro.org> Since the conversion to regmap there has been no need for device level locking for I/O as regmap provides locking so remove the locks. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- drivers/mfd/mc13xxx-core.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/drivers/mfd/mc13xxx-core.c b/drivers/mfd/mc13xxx-core.c index 2a9b100..dbbf8ee 100644 --- a/drivers/mfd/mc13xxx-core.c +++ b/drivers/mfd/mc13xxx-core.c @@ -158,8 +158,6 @@ int mc13xxx_reg_read(struct mc13xxx *mc13xxx, unsigned int offset, u32 *val) { int ret; - BUG_ON(!mutex_is_locked(&mc13xxx->lock)); - if (offset > MC13XXX_NUMREGS) return -EINVAL; @@ -172,8 +170,6 @@ EXPORT_SYMBOL(mc13xxx_reg_read); int mc13xxx_reg_write(struct mc13xxx *mc13xxx, unsigned int offset, u32 val) { - BUG_ON(!mutex_is_locked(&mc13xxx->lock)); - dev_vdbg(mc13xxx->dev, "[0x%02x] <- 0x%06x\n", offset, val); if (offset > MC13XXX_NUMREGS || val > 0xffffff) @@ -186,7 +182,6 @@ EXPORT_SYMBOL(mc13xxx_reg_write); int mc13xxx_reg_rmw(struct mc13xxx *mc13xxx, unsigned int offset, u32 mask, u32 val) { - BUG_ON(!mutex_is_locked(&mc13xxx->lock)); BUG_ON(val & ~mask); dev_vdbg(mc13xxx->dev, "[0x%02x] <- 0x%06x (mask: 0x%06x)\n", offset, val, mask); -- 1.8.4.rc3

2 8

[alsa-devel] [PATCH] ASoC: rt5640: ifdef for ACPI module table
by Olof Johansson 19 Sep '13

19 Sep '13

Commit 02b80773de37 (ASoC: rt5640: Add ACPI probing support.) causes a warning due to unreferenced variable on non-ACPI configs such as tegra_defconfig on ARM: sound/soc/codecs/rt5640.c:2085:30: warning: 'rt5640_acpi_match' defined but not used [-Wunused-variable] Signed-off-by: Olof Johansson <olof(a)lixom.net> --- sound/soc/codecs/rt5640.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 0bfb960..a856706 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2082,11 +2082,13 @@ static const struct i2c_device_id rt5640_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id); +#if CONFIG_ACPI static struct acpi_device_id rt5640_acpi_match[] = { { "INT33CA", 0 }, { }, }; MODULE_DEVICE_TABLE(acpi, rt5640_acpi_match); +#endif static int rt5640_parse_dt(struct rt5640_priv *rt5640, struct device_node *np) { -- 1.7.10.4

3 11

[alsa-devel] [PATCH] ASoC: kirkwood: fix loss of external clock at probe time
by Jean-Francois Moine 19 Sep '13

19 Sep '13

At probe time, when the clock driver is not yet initialized, the external clock of the kirkwood sound device will not be usable. This patch fixes this problem defering the device probe. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/kirkwood/kirkwood-i2s.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 0f3d73d..14f32d9 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -496,7 +496,10 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) return err; priv->extclk = devm_clk_get(&pdev->dev, "extclk"); - if (!IS_ERR(priv->extclk)) { + if (IS_ERR(priv->extclk)) { + if (np && of_property_match_string(np, "clock-names", "extclk") >= 0) + return -EPROBE_DEFER; + } else { if (priv->extclk == priv->clk) { devm_clk_put(&pdev->dev, priv->extclk); priv->extclk = ERR_PTR(-EINVAL); -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 1

Re: [alsa-devel] [PATCH] ASoC: rt5640: Omit ACPI match table only if !ACPI
by Mark Brown 19 Sep '13

19 Sep '13

On Thu, Sep 19, 2013 at 11:18:06AM +0200, Thierry Reding wrote: > The ACPI_PTR() macro evaluates to NULL if ACPI is disabled and hence the > ACPI match table won't be used, causing the compiler to complain. Avoid > this by protecting the table using an #ifdef CONFIG_ACPI. Applied, thanks. Like I was saying to Olof we really should be able to do better than needing the ifdefs though.

1 0

[alsa-devel] [PATCH 1/1] ASoC: SPEAr spdif_out: Remove redundant variable
by Sachin Kamat 19 Sep '13

19 Sep '13

Return directly and remove the intermediate local variable. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/spear/spdif_out.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c index 70fc4d6..b6ef6f7 100644 --- a/sound/soc/spear/spdif_out.c +++ b/sound/soc/spear/spdif_out.c @@ -280,7 +280,6 @@ static int spdif_out_probe(struct platform_device *pdev) struct spdif_out_dev *host; struct spear_spdif_platform_data *pdata; struct resource *res; - int ret; host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); if (!host) { @@ -307,9 +306,8 @@ static int spdif_out_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, host); - ret = devm_snd_soc_register_component(&pdev->dev, &spdif_out_component, - &spdif_out_dai, 1); - return ret; + return devm_snd_soc_register_component(&pdev->dev, &spdif_out_component, + &spdif_out_dai, 1); } #ifdef CONFIG_PM -- 1.7.9.5

2 1

[alsa-devel] Fwd: Re: Realtek ALC889: HDA Intel and kernel 3.1 gives choppy sound (again)
by Colomban Wendling 18 Sep '13

18 Sep '13

Here's my initial mail, but with gzipped outputs. Regards, Colomban -------- Message original -------- Sujet: Re: [alsa-devel] Realtek ALC889: HDA Intel and kernel 3.1 gives choppy sound (again) Date : Thu, 10 Nov 2011 02:11:17 +0100 De : Colomban Wendling <lists.ban(a)herbesfolles.org> Pour : Takashi Iwai <tiwai(a)suse.de> Copie à : alsa-devel(a)alsa-project.org Hi again, and sorry for the delay. Le 08/11/2011 07:32, Takashi Iwai a écrit : > At Mon, 07 Nov 2011 23:45:40 +0100, > Colomban Wendling wrote: >> >> Hi, >> >> Back to 3.0, the sound on my soundcard [1] started to be choppy, I >> reported it [2] and it got fixed (thanks to Takashi Iwai!). >> >> However, the story repeated with 3.1 (and probably 3.0.8 or before [3]): >> I've got similar choppy sound again. >> >> I bisected the few commits that happened on >> sound/pci/hda/patch_realtek.c, and finally found the villain: >> >> "8974bd51 ALSA: hda/realtek - Fix auto-mute with HP+LO configuration" >> >> Reverting it from v3.1 fixes the problem. > > Does it? Very weird. This patch has nothing to do with the HD-audio > controller side but purely a codec change. Yep, I just check once again and it really fixes the problem here. >> master (94956ee) also suffers of the problem, but I couldn't test >> without that commit because reverting it fails, too much changes happened. > > Did you try to pass position_fix=1 (or 2) option? > Also, passing enable_msi=0 (or 1) may change anything? I just tried with the 3 kernels: * vanilla v3.1 * vanilla 3.2-rc1 (master at 1ea6b8f) * v3.1 with 8974bd51 reverted and none of the option changed anything: both vanilla always failed, the third always succeeded. Though, I haven't combined the options, just tried each at once. > In anyway, please give alsa-info.sh output again with the fresh > kernel. Run it with --no-upload option and attach the output. Here it is, for the 3 kernels cited above. Regards, Colomban

4 14

Re: [alsa-devel] snd_pcm_delay issue - How to debug?
by Sander Jansen 18 Sep '13

18 Sep '13

Never mind. Issue not related to alsa. On Tue, Jul 16, 2013 at 6:27 PM, Sander Jansen <s.jansen(a)gmail.com> wrote: > Here the two alsa-info dumps for the hardware I'm having the problem with. > Also a log of my music player. It reports: > > The 'position' the number of frames it has written to the device (also > reported in seconds) and the delay at that time (after writing). > Also the 'timestamp' which is the reported playback progress in seconds > reported to the UI. When the second track is played (line 590), the > problems start to become noticeable eg. playback progress is a couple of > seconds out of sync. > > Any ideas? > > Thanks, > > Sander > > > > On Tue, Jul 16, 2013 at 9:28 AM, Sander Jansen <s.jansen(a)gmail.com> wrote: > >> Hi, >> >> I'm seeing an issue where snd_pcm_delay is returning bogus values >> (usually a lot larger than in actuality) in that it ultimately affects the >> progress timer in my audio player. Both are HDA based audio cards, though >> one connects to the HDMI output and while the other connects to analog out. >> >> I'm suspecting I'm seeing a driver bug since I only start noticing the >> issue since a few kernel releases ago (though not sure which one started >> it). >> >> How can i best debug this issue and find out the cause. >> >> Thanks, >> >> Sander >> > >

1 0

Re: [alsa-devel] [PATCH v2 05/11] ASoC: davinci-mcasp: Remove interrupt property from DT bindings doc
by Mark Brown 18 Sep '13

18 Sep '13

On Wed, Sep 18, 2013 at 09:48:34AM -0500, Nishanth Menon wrote: > On 09/18/2013 09:46 AM, Mark Brown wrote: > > How about moving these to optional instead of removing them completely > > (or just leaving them alone)? That way if the driver does want to start > > using them people will have hooked them up already so no updates will be > > required. > the definitions as they stand now make no sense as the ordering of the > interrupts are undefined - example is the first index RX or TX? the > alternative will be PIO McASP driver, which might be nice, but having > a vague binding with 0 code behind is of no use for anyone right? So the sensible fix seems to be to specify the order rather than remove them then...

1 0

[alsa-devel] [PATCH 1/2] ASoC: SPEAr spdif_in: Use devm_snd_soc_register_component
by Sachin Kamat 18 Sep '13

18 Sep '13

devm_snd_soc_register_component makes code simpler. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/spear/spdif_in.c | 12 ++---------- 1 file changed, 2 insertions(+), 10 deletions(-) diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c index 63acfeb..21a8c95 100644 --- a/sound/soc/spear/spdif_in.c +++ b/sound/soc/spear/spdif_in.c @@ -257,20 +257,12 @@ static int spdif_in_probe(struct platform_device *pdev) return ret; } - return snd_soc_register_component(&pdev->dev, &spdif_in_component, - &spdif_in_dai, 1); -} - -static int spdif_in_remove(struct platform_device *pdev) -{ - snd_soc_unregister_component(&pdev->dev); - - return 0; + return devm_snd_soc_register_component(&pdev->dev, &spdif_in_component, + &spdif_in_dai, 1); } static struct platform_driver spdif_in_driver = { .probe = spdif_in_probe, - .remove = spdif_in_remove, .driver = { .name = "spdif-in", .owner = THIS_MODULE, -- 1.7.9.5

3 5

[alsa-devel] Incorrect sbits for snd-usb-audio
by Sander Jansen 17 Sep '13

17 Sep '13

Hi, I have a Focusrite Scarlett 2i4 USB Audio Interface. The audio format exposed by ALSA for this device is S32_LE. From the hardware specs and lsusb. it supposed to only have a 24 bit resolution (with the LSB bits set to 0): AudioStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bSubslotSize 4 bBitResolution 24 When I query the sbits for this device, I expect it to return 24, but instead it returns 32. Looking through the usb audio driver code, it doesn't look like this information is actually exposed to userspace. The 'parse_audio_format_i_type' in usb/format.c merely returns the supported formats, but doesn't include any msbits info. (and it doesn't look like it can be easily changed to return the msbits info as well). Was this intentionally left out, or is this a missing feature in the driver? Thanks, Sander

1 0

[alsa-devel] [PATCH 0/4] add regmap_filelds and use it on Renesas Sound driver
by Kuninori Morimoto 17 Sep '13

17 Sep '13

Hi Mark These patches add new regmap_filelds API on kernel. ~ and use it on Renesas sound driver instead of original method. It can care about multi port register offset via regmap. 0xAAAA + 0x40 -- port 0 -- regX regY regZ 0xAAAA + 0x80 -- port 1 -- regX regY regZ This case, current API needs 2 (= port) x 3 (= regX/Y/Z) regmap_fileld, but this new API can care about all port via 3 regmap_filelds with port ID. I'm not sure that regmap_filelds is good naming or not. Please let me know if you have good naming idea. These are based on below branchs regmap/for-next + asoc/for-next Kuninori Morimoto (4): regmap: add regmap_field_update_bits() regmap: Add regmap_fields APIs ASoC: rsnd: gen: rsnd_gen_ops cares .probe and .remove ASoC: rsnd: use regmap instead of original register mapping method drivers/base/regmap/internal.h | 3 + drivers/base/regmap/regmap.c | 117 +++++++++++++++++ include/linux/regmap.h | 14 +++ sound/soc/sh/rcar/core.c | 45 ------- sound/soc/sh/rcar/gen.c | 270 +++++++++++++++++++++++++--------------- 5 files changed, 307 insertions(+), 142 deletions(-) Best regards --- Kuninori Morimoto

2 14

[alsa-devel] [PATCH 1/1] ASoC: fsl_ssi: Remove redundant dev_set_drvdata
by Sachin Kamat 17 Sep '13

17 Sep '13

Driver core sets the driver data to NULL on detach. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/fsl/fsl_ssi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 4973be7..6ac8730 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1135,7 +1135,6 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (ssi_private->ssi_on_imx) imx_pcm_dma_exit(pdev); snd_soc_unregister_component(&pdev->dev); - dev_set_drvdata(&pdev->dev, NULL); device_remove_file(&pdev->dev, &ssi_private->dev_attr); if (ssi_private->ssi_on_imx) clk_disable_unprepare(ssi_private->clk); -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/1] ASoC: fsl_spdif: Remove redundant semicolon
by Sachin Kamat 17 Sep '13

17 Sep '13

Redundant semicolon at the end of brace is removed. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/fsl/fsl_spdif.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 3920c3e..c0fea02 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -963,7 +963,7 @@ static bool fsl_spdif_readable_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static bool fsl_spdif_writeable_reg(struct device *dev, unsigned int reg) @@ -982,7 +982,7 @@ static bool fsl_spdif_writeable_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static const struct regmap_config fsl_spdif_regmap_config = { -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/1] ASoC: ak4642: Remove redundant break
by Sachin Kamat 17 Sep '13

17 Sep '13

'break' after return statement is redundant. Remove it. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/codecs/ak4642.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 2d03787..21c35ed 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -352,7 +352,6 @@ static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) */ default: return -EINVAL; - break; } snd_soc_update_bits(codec, MD_CTL1, DIF_MASK, data); @@ -405,7 +404,6 @@ static int ak4642_dai_hw_params(struct snd_pcm_substream *substream, break; default: return -EINVAL; - break; } snd_soc_update_bits(codec, MD_CTL2, FS_MASK, rate); -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/1] ASoC: twl6040: Remove redundant semicolon
by Sachin Kamat 17 Sep '13

17 Sep '13

Redundant semicolon removed. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/codecs/twl6040.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 3c79dbb..35059a2 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -246,7 +246,7 @@ static bool twl6040_is_path_unmuted(struct snd_soc_codec *codec, return priv->dl2_unmuted; default: return 1; - }; + } } /* @@ -1100,7 +1100,7 @@ static void twl6040_mute_path(struct snd_soc_codec *codec, enum twl6040_dai_id i break; default: break; - }; + } } static int twl6040_digital_mute(struct snd_soc_dai *dai, int mute) -- 1.7.9.5

3 2

Re: [alsa-devel] [PATCH] ASoC: blackfin: Add missing break statement to bf6xx
by Mark Brown 17 Sep '13

17 Sep '13

On Sat, Sep 14, 2013 at 02:20:37AM +0300, Valentin Ilie wrote: > SNDRV_PCM_FORMAT_S8 isn't supposed to fall through to SNDRV_PCM_FORMAT_S16_LE > > Signed-off-by: Valentin Ilie <valentin.ilie(a)gmail.com> Applied, thanks.

1 0

[alsa-devel] [PATCH] ASoC: ak4104: provide a module device table
by Daniel Mack 17 Sep '13

17 Sep '13

Provide a module device table for the SPI subsystem, so the driver can be autoloaded by the SPI core. While at it, get rid of an unnecessary #define. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/codecs/ak4104.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index 71059c0..b4819dc 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -45,8 +45,6 @@ #define AK4104_TX_TXE (1 << 0) #define AK4104_TX_V (1 << 1) -#define DRV_NAME "ak4104-codec" - struct ak4104_private { struct regmap *regmap; }; @@ -291,12 +289,19 @@ static const struct of_device_id ak4104_of_match[] = { }; MODULE_DEVICE_TABLE(of, ak4104_of_match); +static const struct spi_device_id ak4104_id_table[] = { + { "ak4104", 0 }, + { } +}; +MODULE_DEVICE_TABLE(spi, ak4104_id_table); + static struct spi_driver ak4104_spi_driver = { .driver = { - .name = DRV_NAME, + .name = "ak4104", .owner = THIS_MODULE, .of_match_table = ak4104_of_match, }, + .id_table = ak4104_id_table, .probe = ak4104_spi_probe, .remove = ak4104_spi_remove, }; -- 1.8.3.1

2 1

[alsa-devel] [PATCH] ASoC: arizona: Improve handling of setting REFCLK to 0
by Charles Keepax 17 Sep '13

17 Sep '13

This patch suppresses calculation of REFCLK parameters when the REFCLK source frequency is set to zero, additionally it will consider a source frequency of zero as the REFCLK being disabled and switch to using the SYNCCLK. Reported-by: Kyung Kwee Ryu <Kyung-Kwee.Ryu(a)wolfsonmicro.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- This patch missed getting upstreamed at the time it was written, sorry about that. Thanks, Charles sound/soc/codecs/arizona.c | 23 ++++++++++++++--------- 1 files changed, 14 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 657808b..6f05b17 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1477,21 +1477,25 @@ static void arizona_enable_fll(struct arizona_fll *fll, { struct arizona *arizona = fll->arizona; int ret; + bool use_sync = false; /* * If we have both REFCLK and SYNCCLK then enable both, * otherwise apply the SYNCCLK settings to REFCLK. */ - if (fll->ref_src >= 0 && fll->ref_src != fll->sync_src) { + if (fll->ref_src >= 0 && fll->ref_freq && + fll->ref_src != fll->sync_src) { regmap_update_bits(arizona->regmap, fll->base + 5, ARIZONA_FLL1_OUTDIV_MASK, ref->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); arizona_apply_fll(arizona, fll->base, ref, fll->ref_src, false); - if (fll->sync_src >= 0) + if (fll->sync_src >= 0) { arizona_apply_fll(arizona, fll->base + 0x10, sync, fll->sync_src, true); + use_sync = true; + } } else if (fll->sync_src >= 0) { regmap_update_bits(arizona->regmap, fll->base + 5, ARIZONA_FLL1_OUTDIV_MASK, @@ -1511,7 +1515,7 @@ static void arizona_enable_fll(struct arizona_fll *fll, * Increase the bandwidth if we're not using a low frequency * sync source. */ - if (fll->sync_src >= 0 && fll->sync_freq > 100000) + if (use_sync && fll->sync_freq > 100000) regmap_update_bits(arizona->regmap, fll->base + 0x17, ARIZONA_FLL1_SYNC_BW, 0); else @@ -1526,8 +1530,7 @@ static void arizona_enable_fll(struct arizona_fll *fll, regmap_update_bits(arizona->regmap, fll->base + 1, ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA); - if (fll->ref_src >= 0 && fll->sync_src >= 0 && - fll->ref_src != fll->sync_src) + if (use_sync) regmap_update_bits(arizona->regmap, fll->base + 0x11, ARIZONA_FLL1_SYNC_ENA, ARIZONA_FLL1_SYNC_ENA); @@ -1561,10 +1564,12 @@ int arizona_set_fll_refclk(struct arizona_fll *fll, int source, if (fll->ref_src == source && fll->ref_freq == Fref) return 0; - if (fll->fout && Fref > 0) { - ret = arizona_calc_fll(fll, &ref, Fref, fll->fout); - if (ret != 0) - return ret; + if (fll->fout) { + if (Fref > 0) { + ret = arizona_calc_fll(fll, &ref, Fref, fll->fout); + if (ret != 0) + return ret; + } if (fll->sync_src >= 0) { ret = arizona_calc_fll(fll, &sync, fll->sync_freq, -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ASoC: bells: Add missing route to power up DSP clock
by Charles Keepax 17 Sep '13

17 Sep '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/bells.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/samsung/bells.c b/sound/soc/samsung/bells.c index 29e2468..84f5d8b 100644 --- a/sound/soc/samsung/bells.c +++ b/sound/soc/samsung/bells.c @@ -356,6 +356,7 @@ static struct snd_soc_dapm_widget bells_widgets[] = { static struct snd_soc_dapm_route bells_routes[] = { { "Sub CLK_SYS", NULL, "OPCLK" }, + { "CLKIN", NULL, "OPCLK" }, { "DMIC", NULL, "MICBIAS2" }, { "IN2L", NULL, "DMIC" }, -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ASoC: core: Add API for configuration of DAI BCLK ratio
by Liam Girdwood 17 Sep '13

17 Sep '13

Some codec drivers when running in slave mode require that BCLK to sample rate ratio is explicitly set by the machine driver as it may not be exactly rate * frame size. Extend the DAI API by adding :- int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio); Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- include/sound/soc-dai.h | 3 +++ sound/soc/soc-core.c | 16 ++++++++++++++++ 2 files changed, 19 insertions(+) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index ae9a227..d8acf0c 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -105,6 +105,8 @@ int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out); +int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio); + /* Digital Audio interface formatting */ int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); @@ -131,6 +133,7 @@ struct snd_soc_dai_ops { int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out); int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div); + int (*set_bclk_ratio)(struct snd_soc_dai *dai, unsigned int ratio); /* * DAI format configuration diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 09e341d..8014839 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3380,6 +3380,22 @@ int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source, EXPORT_SYMBOL_GPL(snd_soc_codec_set_pll); /** + * snd_soc_dai_set_bclk_ratio - configure BCLK to sample rate ratio. + * @dai: DAI + * @ratio Ratio of BCLK to Sample rate. + * + * Configures the DAI for a preset BCLK to sample rate ratio. + */ +int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + if (dai->driver && dai->driver->ops->set_bclk_ratio) + return dai->driver->ops->set_bclk_ratio(dai, ratio); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_bclk_ratio); + +/** * snd_soc_dai_set_fmt - configure DAI hardware audio format. * @dai: DAI * @fmt: SND_SOC_DAIFMT_ format value. -- 1.8.1.2

2 1

[alsa-devel] USB Audio initialization race
by David Henningsson 17 Sep '13

17 Sep '13

I'm sometimes seeing that USB audio devices are not initialized correctly by PulseAudio, but I can't reproduce this myself. I have, however, spotted what I think is a race condition in the driver code, if a device has more than one interface. snd_card_register is called once for every interface, which means that it will be shown to userspace after the first interface has been probed. So the race could look like: 1) The first interface is probed, which might contain the mixer controls 2) snd_card_register is called, which sends a signal to userspace 3) PulseAudio probes the device and notices that it has no PCMs, so it ignores the device 4) The second interface is probed, which adds the PCMs 5) snd_card_register is again called, which is a no-op since the card already exist 6) User is unhappy because his plugged in sound card did not show up in PulseAudio. So, assuming all this is right, it seems like we need some type of callback from the usb driver when all the interfaces for the sound card has been probed, so we can call snd_card_register at that point instead. Thoughts? -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

3 10

[alsa-devel] [PATCH 1/2] ASoC: core: Add devm_snd_soc_register_card()
by Mark Brown 16 Sep '13

16 Sep '13

From: Mark Brown <broonie(a)linaro.org> Simplify error handling and remove repetitive (and rarely executed) code for unregistration by providing a devm_snd_soc_register() card. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- include/sound/soc.h | 1 + sound/soc/soc-devres.c | 34 ++++++++++++++++++++++++++++++++++ 2 files changed, 35 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index f126676..7ab5d2b 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -364,6 +364,7 @@ int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source, int snd_soc_register_card(struct snd_soc_card *card); int snd_soc_unregister_card(struct snd_soc_card *card); +int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card); int snd_soc_suspend(struct device *dev); int snd_soc_resume(struct device *dev); int snd_soc_poweroff(struct device *dev); diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c index 13fe86f..b1d7322 100644 --- a/sound/soc/soc-devres.c +++ b/sound/soc/soc-devres.c @@ -50,3 +50,37 @@ int devm_snd_soc_register_component(struct device *dev, return ret; } EXPORT_SYMBOL_GPL(devm_snd_soc_register_component); + +static void devm_card_release(struct device *dev, void *res) +{ + snd_soc_unregister_card(*(struct snd_soc_card **)res); +} + +/** + * devm_snd_soc_register_card - resource managed card registration + * @dev: Device used to manage card + * @card: Card to register + * + * Register a card with automatic unregistration when the device is + * unregistered. + */ +int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card) +{ + struct device **ptr; + int ret; + + ptr = devres_alloc(devm_card_release, sizeof(*ptr), GFP_KERNEL); + if (!ptr) + return -ENOMEM; + + ret = snd_soc_register_card(card); + if (ret == 0) { + *ptr = dev; + devres_add(dev, ptr); + } else { + devres_free(ptr); + } + + return ret; +} +EXPORT_SYMBOL_GPL(devm_snd_soc_register_card); -- 1.8.4.rc3

2 2

[alsa-devel] [PATCH] ASoC: core: Only add platform DAI widgets once.
by Liam Girdwood 16 Sep '13

16 Sep '13

Currently platform CPU DAI widgets are created in soc_probe_platform and soc_probe_link_dais. Remove the extra call in soc_probe_link_dais(). Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/soc-core.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 8e86aba..09e341d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1365,7 +1365,6 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) return -ENODEV; list_add(&cpu_dai->dapm.list, &card->dapm_list); - snd_soc_dapm_new_dai_widgets(&cpu_dai->dapm, cpu_dai); } if (cpu_dai->driver->probe) { -- 1.8.1.2

2 10

Re: [alsa-devel] No sound from internal speaker on 2013 Macbook Air 6, 1
by Ben Whitten 16 Sep '13

16 Sep '13

> > > OK, then the normal jack detection doesn't work with this machine, as > > > it seems. You can try the above to all pins for checking if any of > > > other pins reacts to the headphone jack. > > > > No luck I'm afraid - none of the other pins seem to react to the > > headphones. Is there anything else you can think of that I can try > > tomorrow? > > Well, it might be notified via GPIO. Try to read GPIO pins except for > GPIO0. That is, set the all unmasked, the direction to READ. We have found that jack sense comes in on the headphone pin in a windows bootcamp install. However this bit isn't set in linux using either the default chip, bootloader, or windows pin configs. No other GPIO's are enabled in windows. Ben

1 0

[alsa-devel] Roland UA-55 (snd-usb-audio)
by Michael Findlater 15 Sep '13

15 Sep '13

Hello there! I am attempting to use the Roland UA-55, which apparently should work after recent changes were made for support for Roland devices by Clemens Ladisch. I am using the 3.11 kernel upon Debian sid. The device shows in /proc/asound/cards but does not seem to work. The GNOME volume mixer it shows the device as a "QUAD-CAPTURE Analog Surround 4.0", but I am not sure whether this is correct. When I select the card as the output device and try to play to it, nothing will play, 'aplay' does not work either (I hear nothing). If I turn the headphone output knob on the device up I hear a very soft static in my right ear. 'alsamixer' reports that 'This sound device does not have any controls.' I do not know where to go from here, but have ran the "Alsa Information Script", an output of this is available at: http://www.alsa-project.org/db/?f=8f62ce460ecc5cac394c0b014b2e2aa4d28ae2dc I am stuck with how to proceed to debug the issue. If at all possible, I would be grateful if someone is able to point me in the right direction. Best regards, Michael Findlater

1 0

[alsa-devel] [PATCH RFC 0/8] Fix AM335x-evm analog audio support
by oku＠iki.fi 13 Sep '13

13 Sep '13

From: Jyri Sarha <jsarha(a)ti.com> This set of patches fixes the basic audio support for am335x-evm. It should also be relatively simple to add the necessary nodes to relevant dts files to get BeagleBone + AudioCape and am335x-evmsk working too. The patch set depends on following patches: [PATCH v2] ARM: dts: add AM33XX EDMA support https://lkml.org/lkml/2013/8/26/57 [PATCH v11 5/8] ARM: dts: add AM33XX SPI DMA support https://lkml.org/lkml/2013/6/18/55 [PATCH v2] ARM: EDMA: Fix clearing of unused list for DT DMA resources https://lkml.org/lkml/2013/7/22/441 I have tried my best not to break the existing support for older davinci boards, but since I do not have those boards I can not be sure. Some commit comments refer to a dmaengine based davinci audio implementation which is planned for but nothing has been done yet. Best regards, Jyri Darren Etheridge (1): ARM/dts: am335x-evm: Add audio support for am335x-evm.dts Hebbar, Gururaja (2): ASoC: davinci-evm: Add device tree binding ASoC: davinci: Add support for AM33xx SoC Audio Joel A Fernandes (1): ASoC: davinci-mcasp: Add pinctrl support Jyri Sarha (3): ASoC: davinci-mcasp: Get DMA related properties from DT ASoC: davinci-mcasp: Extract DMA channels directly from DT ARM/dts: am33xx: mcasp: Add new dma related properties Pantelis Antoniou (1): ARM/dts: am33xx: Add mcasp0 and mcasp1 device tree entries .../bindings/sound/davinci-evm-audio.txt | 53 ++++++ .../bindings/sound/davinci-mcasp-audio.txt | 13 +- arch/arm/boot/dts/am335x-evm.dts | 56 ++++++ arch/arm/boot/dts/am33xx.dtsi | 35 ++++ include/linux/platform_data/davinci_asp.h | 2 + sound/soc/davinci/Kconfig | 8 + sound/soc/davinci/Makefile | 3 + sound/soc/davinci/davinci-evm.c | 181 +++++++++++++++++--- sound/soc/davinci/davinci-mcasp.c | 82 +++++++-- 9 files changed, 391 insertions(+), 42 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/davinci-evm-audio.txt -- 1.7.9.5

4 25

[alsa-devel] [patch] ASoC: ab8500-codec: info leak in anc_status_control_put()
by Dan Carpenter 13 Sep '13

13 Sep '13

If the user passes an invalid value it leads to an info leak when we print the error message or it could oops. This is called with user supplied data from snd_ctl_elem_write(). Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index b8ba0ad..80555d7 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -1225,13 +1225,18 @@ static int anc_status_control_put(struct snd_kcontrol *kcontrol, struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev); struct device *dev = codec->dev; bool apply_fir, apply_iir; - int req, status; + unsigned int req; + int status; dev_dbg(dev, "%s: Enter.\n", __func__); mutex_lock(&drvdata->anc_lock); req = ucontrol->value.integer.value[0]; + if (req >= ARRAY_SIZE(enum_anc_state)) { + status = -EINVAL; + goto cleanup; + } if (req != ANC_APPLY_FIR_IIR && req != ANC_APPLY_FIR && req != ANC_APPLY_IIR) { dev_err(dev, "%s: ERROR: Unsupported status to set '%s'!\n",

2 1

[alsa-devel] [patch] ASoC: 88pm860x: array overflow in snd_soc_put_volsw_2r_st()
by Dan Carpenter 13 Sep '13

13 Sep '13

This is called from snd_ctl_elem_write() with user supplied data so we need to add some bounds checking. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 8af0434..259d1ac 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -349,6 +349,9 @@ static int snd_soc_put_volsw_2r_st(struct snd_kcontrol *kcontrol, val = ucontrol->value.integer.value[0]; val2 = ucontrol->value.integer.value[1]; + if (val >= ARRAY_SIZE(st_table) || val2 >= ARRAY_SIZE(st_table)) + return -EINVAL; + err = snd_soc_update_bits(codec, reg, 0x3f, st_table[val].m); if (err < 0) return err;

2 1

[alsa-devel] [patch] ASoC: max98095: a couple array underflows
by Dan Carpenter 13 Sep '13

13 Sep '13

The ->put() function are called from snd_ctl_elem_write() with user supplied data. The limit checks here could underflow leading to a crash. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 41cdd16..8dbcacd 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -1863,7 +1863,7 @@ static int max98095_put_eq_enum(struct snd_kcontrol *kcontrol, struct max98095_pdata *pdata = max98095->pdata; int channel = max98095_get_eq_channel(kcontrol->id.name); struct max98095_cdata *cdata; - int sel = ucontrol->value.integer.value[0]; + unsigned int sel = ucontrol->value.integer.value[0]; struct max98095_eq_cfg *coef_set; int fs, best, best_val, i; int regmask, regsave; @@ -2016,7 +2016,7 @@ static int max98095_put_bq_enum(struct snd_kcontrol *kcontrol, struct max98095_pdata *pdata = max98095->pdata; int channel = max98095_get_bq_channel(codec, kcontrol->id.name); struct max98095_cdata *cdata; - int sel = ucontrol->value.integer.value[0]; + unsigned int sel = ucontrol->value.integer.value[0]; struct max98095_biquad_cfg *coef_set; int fs, best, best_val, i; int regmask, regsave;

2 1

[alsa-devel] [PATCH] ASoC: rt5640: Provide more useful hw_params error reasons.
by Liam Girdwood 13 Sep '13

13 Sep '13

Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/codecs/rt5640.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 2f6bb16..de40bd9 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1609,7 +1609,8 @@ static int rt5640_hw_params(struct snd_pcm_substream *substream, rt5640->lrck[dai->id] = params_rate(params); pre_div = get_clk_info(rt5640->sysclk, rt5640->lrck[dai->id]); if (pre_div < 0) { - dev_err(codec->dev, "Unsupported clock setting\n"); + dev_err(codec->dev, "Unsupported clock setting %d for DAI %d\n", + rt5640->lrck[dai->id], dai->id); return -EINVAL; } frame_size = snd_soc_params_to_frame_size(params); -- 1.8.1.2

2 2

[alsa-devel] virtual alsa mixer device
by André Roth 13 Sep '13

13 Sep '13

hello, I'd like to implement a mixer control device for an external amplifier, which can be controlled via rs232. would be nice to adjust volume with any alsa mixer program.. is there a way to do this in user space ? any hints for the right api / examples ? -- andré

1 0

[alsa-devel] [PATCH] ASoC: codecs: da9055: Update driver name to fix breakage due to pmic driver with same name
by Ashish Chavan 13 Sep '13

13 Sep '13

This patch updates i2c driver name and device id of da9055 codec driver. DA9055 is a PMIC + CODEC and currently, the corresponding PMIC driver also registers itself with the same name as codec, i.e. "da9055". Because of this the codec driver was broken. Now codec driver uses "da9055-codec" as driver name instead of "da9055". Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <david.chen(a)diasemi.com> --- sound/soc/codecs/da9055.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c index 2e1bf5d..e146f68 100644 --- a/sound/soc/codecs/da9055.c +++ b/sound/soc/codecs/da9055.c @@ -1544,7 +1544,7 @@ static int da9055_remove(struct i2c_client *client) } static const struct i2c_device_id da9055_i2c_id[] = { - { "da9055", 0 }, + { "da9055-codec", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, da9055_i2c_id); @@ -1552,7 +1552,7 @@ MODULE_DEVICE_TABLE(i2c, da9055_i2c_id); /* I2C codec control layer */ static struct i2c_driver da9055_i2c_driver = { .driver = { - .name = "da9055", + .name = "da9055-codec", .owner = THIS_MODULE, }, .probe = da9055_i2c_probe, -- 1.7.1

3 29

[alsa-devel] No sound from internal speaker on 2013 Macbook Air 6, 1
by Ian Munsie 13 Sep '13

13 Sep '13

Hey all, I'm trying to get sound working on my new mid 2013 Macbook Air 6,1. Sound works via the headphone jack, but I haven't managed to get the internal speaker to work. I've checked all the controls in the mixer, tried a few different mac related model= options to snd-hda-intel and even tried tweaking things with hda_analyzer to no avail. I've uninstalled pulseaudio (for now) to get rid of it's ALSA hook, but had the same result both with and without it. I'm running on a 3.11-rc6+ built today (had a few other fixes to make for this machine) - I didn't bother including the hda i915 driver as I don't need sound via the display port, but the hda driver does print a warning about that missing module. The speaker wasn't working on the 3.10 kernel from Debian either which included that module. Any help would be appreciated - I've attached my alsa-info output taken after a reboot. Cheers, -Ian -- http://sites.google.com/site/DarkStarJunkSpace -- http://darkstarshout.blogspot.com/ -- Please avoid sending me Word or PowerPoint attachments. See http://www.gnu.org/philosophy/no-word-attachments.html

3 15

[alsa-devel] [PATCH 1/1] ALSA: rme9652: Remove redundant break
by Sachin Kamat 13 Sep '13

13 Sep '13

'break' after return statement is not necessary. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/pci/rme9652/hdspm.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 3cde55b..2907e68 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -3996,7 +3996,6 @@ static int hdspm_tco_sync_check(struct hdspm *hdspm) return 1; } return 0; - break; case AES32: status = hdspm_read(hdspm, HDSPM_statusRegister); if (status & HDSPM_tcoLockAes) { @@ -4006,9 +4005,6 @@ static int hdspm_tco_sync_check(struct hdspm *hdspm) return 1; } return 0; - - break; - case RayDAT: case AIO: status = hdspm_read(hdspm, HDSPM_RD_STATUS_1); @@ -4018,7 +4014,6 @@ static int hdspm_tco_sync_check(struct hdspm *hdspm) if (status & 0x4000000) return 1; /* Lock */ return 0; /* No signal */ - break; default: break; -- 1.7.9.5

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[alsa-devel] [PATCH 1/1] ALSA: au88x0: Remove redundant break
by Sachin Kamat 13 Sep '13

13 Sep '13

'break' after a return statement is redundant. Remove it. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/pci/au88x0/au88x0_synth.c | 29 ++++++++++------------------- 1 file changed, 10 insertions(+), 19 deletions(-) diff --git a/sound/pci/au88x0/au88x0_synth.c b/sound/pci/au88x0/au88x0_synth.c index 8bef473..922a84b 100644 --- a/sound/pci/au88x0/au88x0_synth.c +++ b/sound/pci/au88x0/au88x0_synth.c @@ -219,7 +219,6 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, */ hwwrite(vortex->mmio, WT_RUN(wt), val); return 0xc; - break; case 1: /* param 0 */ /* printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", @@ -227,7 +226,6 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, */ hwwrite(vortex->mmio, WT_PARM(wt, 0), val); return 0xc; - break; case 2: /* param 1 */ /* printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", @@ -235,7 +233,6 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, */ hwwrite(vortex->mmio, WT_PARM(wt, 1), val); return 0xc; - break; case 3: /* param 2 */ /* printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", @@ -243,7 +240,6 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, */ hwwrite(vortex->mmio, WT_PARM(wt, 2), val); return 0xc; - break; case 4: /* param 3 */ /* printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", @@ -251,7 +247,6 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, */ hwwrite(vortex->mmio, WT_PARM(wt, 3), val); return 0xc; - break; case 6: /* mute */ /* printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", @@ -259,20 +254,17 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, */ hwwrite(vortex->mmio, WT_MUTE(wt), val); return 0xc; - break; case 0xb: - { /* delay */ - /* - printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", - WT_DELAY(wt,0), (int)val); - */ - hwwrite(vortex->mmio, WT_DELAY(wt, 3), val); - hwwrite(vortex->mmio, WT_DELAY(wt, 2), val); - hwwrite(vortex->mmio, WT_DELAY(wt, 1), val); - hwwrite(vortex->mmio, WT_DELAY(wt, 0), val); - return 0xc; - } - break; + /* delay */ + /* + printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", + WT_DELAY(wt,0), (int)val); + */ + hwwrite(vortex->mmio, WT_DELAY(wt, 3), val); + hwwrite(vortex->mmio, WT_DELAY(wt, 2), val); + hwwrite(vortex->mmio, WT_DELAY(wt, 1), val); + hwwrite(vortex->mmio, WT_DELAY(wt, 0), val); + return 0xc; /* Global WT block parameters */ case 5: /* sramp */ ecx = WT_SRAMP(wt); @@ -291,7 +283,6 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, break; default: return 0; - break; } /* printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", ecx, (int)val); -- 1.7.9.5

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[alsa-devel] [PATCH 1/2] ALSA: ctxfi: Staticize local symbols
by Sachin Kamat 13 Sep '13

13 Sep '13

Local symbols used only in this file are made static. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/pci/ctxfi/ctdaio.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ctxfi/ctdaio.c b/sound/pci/ctxfi/ctdaio.c index 0c00eb4..84f86bf 100644 --- a/sound/pci/ctxfi/ctdaio.c +++ b/sound/pci/ctxfi/ctdaio.c @@ -33,7 +33,7 @@ struct daio_rsc_idx { unsigned short right; }; -struct daio_rsc_idx idx_20k1[NUM_DAIOTYP] = { +static struct daio_rsc_idx idx_20k1[NUM_DAIOTYP] = { [LINEO1] = {.left = 0x00, .right = 0x01}, [LINEO2] = {.left = 0x18, .right = 0x19}, [LINEO3] = {.left = 0x08, .right = 0x09}, @@ -44,7 +44,7 @@ struct daio_rsc_idx idx_20k1[NUM_DAIOTYP] = { [SPDIFI1] = {.left = 0x95, .right = 0x9d}, }; -struct daio_rsc_idx idx_20k2[NUM_DAIOTYP] = { +static struct daio_rsc_idx idx_20k2[NUM_DAIOTYP] = { [LINEO1] = {.left = 0x40, .right = 0x41}, [LINEO2] = {.left = 0x60, .right = 0x61}, [LINEO3] = {.left = 0x50, .right = 0x51}, -- 1.7.9.5

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[alsa-devel] [patch] ALSA: asihpi: a couple array out of bounds issues
by Dan Carpenter 13 Sep '13

13 Sep '13

These ->put() functions are called from snd_ctl_elem_write() with user supplied data. snd_asihpi_tuner_band_put() is missing a limit check and the check in snd_asihpi_clksrc_put() can underflow. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index dc632cd..5f2acd3 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -1913,6 +1913,7 @@ static int snd_asihpi_tuner_band_put(struct snd_kcontrol *kcontrol, struct snd_card_asihpi *asihpi = snd_kcontrol_chip(kcontrol); */ u32 h_control = kcontrol->private_value; + unsigned int idx; u16 band; u16 tuner_bands[HPI_TUNER_BAND_LAST]; u32 num_bands = 0; @@ -1920,7 +1921,10 @@ static int snd_asihpi_tuner_band_put(struct snd_kcontrol *kcontrol, num_bands = asihpi_tuner_band_query(kcontrol, tuner_bands, HPI_TUNER_BAND_LAST); - band = tuner_bands[ucontrol->value.enumerated.item[0]]; + idx = ucontrol->value.enumerated.item[0]; + if (idx >= ARRAY_SIZE(tuner_bands)) + idx = ARRAY_SIZE(tuner_bands) - 1; + band = tuner_bands[idx]; hpi_handle_error(hpi_tuner_set_band(h_control, band)); return 1; @@ -2383,7 +2387,8 @@ static int snd_asihpi_clksrc_put(struct snd_kcontrol *kcontrol, struct snd_card_asihpi *asihpi = (struct snd_card_asihpi *)(kcontrol->private_data); struct clk_cache *clkcache = &asihpi->cc; - int change, item; + unsigned int item; + int change; u32 h_control = kcontrol->private_value; change = 1;

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[alsa-devel] [PATCH 1/2] ASoC: fsl_ssi: Staticize local symbols
by Sachin Kamat 13 Sep '13

13 Sep '13

Local symbols used only in this file are made static. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/fsl/fsl_ssi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c6b7439..4973be7 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -748,7 +748,7 @@ static void fsl_ssi_ac97_init(void) fsl_ssi_setup(fsl_ac97_data); } -void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, +static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) { struct ccsr_ssi *ssi = fsl_ac97_data->ssi; @@ -770,7 +770,7 @@ void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, udelay(100); } -unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, +static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, unsigned short reg) { struct ccsr_ssi *ssi = fsl_ac97_data->ssi; -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: SAMSUNG: Allow mono in i2s driver
by Charles Keepax 13 Sep '13

13 Sep '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- I can't see any reason to not allow mono playback in the i2s driver and indeed it seems to work well on our test system here. However, I am fairly unfamilar with the code in this part of the tree so apologies if I am missing some constraint. Thanks, Charles sound/soc/samsung/i2s.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 959c702..70d08d7 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -991,7 +991,7 @@ static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) i2s->i2s_dai_drv.ops = &samsung_i2s_dai_ops; i2s->i2s_dai_drv.suspend = i2s_suspend; i2s->i2s_dai_drv.resume = i2s_resume; - i2s->i2s_dai_drv.playback.channels_min = 2; + i2s->i2s_dai_drv.playback.channels_min = 1; i2s->i2s_dai_drv.playback.channels_max = 2; i2s->i2s_dai_drv.playback.rates = SAMSUNG_I2S_RATES; i2s->i2s_dai_drv.playback.formats = SAMSUNG_I2S_FMTS; -- 1.7.2.5

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[alsa-devel] USB Audio/ALSA
by Rajaram R 13 Sep '13

13 Sep '13

Hi I am trying to bring up Android Accessory Audio in my Samsung Grand mobile.. I am using a Linux PC for the same. The mobile re-enumerates in AOA mode with an Audio interface. But when I play song, I see nothing on the bus. I can see an AOA audio device entry in /proc/asound/cards. The mobile runs Android JB The error i see when I play is "Sorry this player does not support this type of audio file android" . I also tried a Note 2 & tab 2 with JB, i dont see the error , but still i dont see anything on the bus.. Nexus is working fine in the same setup. Also logcat shows information as below. When i search net i understand some problem with ALSA/Android handshake AudioTrack: obtainBuffer timed out (is the CPU pegged?).......some address............ AudioTrack: obtainBuffer timed out (is the CPU pegged?).......some address............ AudioTrack: obtainBuffer timed out (is the CPU pegged?).......some address............ AudioTrack: obtainBuffer timed out (is the CPU pegged?).......some address............ Looking forward for some support

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[alsa-devel] [PATCH 1/2] snd-ac97-codec: Add ID for TI TLV320AIC27 codec
by Ondrej Zary 12 Sep '13

12 Sep '13

Add 0x54584e03 ID for TI TLV320AIC27 AC'97 codec according to datasheet: http://www.ti.com/lit/ds/slas253a/slas253a.pdf The weird thing is that the chip is physically marked 320AD91. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- sound/pci/ac97/ac97_codec.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 445ca48..bf578ba2 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -175,6 +175,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x54524106, 0xffffffff, "TR28026", NULL, NULL }, { 0x54524108, 0xffffffff, "TR28028", patch_tritech_tr28028, NULL }, // added by xin jin [07/09/99] { 0x54524123, 0xffffffff, "TR28602", NULL, NULL }, // only guess --jk [TR28023 = eMicro EM28023 (new CT1297)] +{ 0x54584e03, 0xffffffff, "TLV320AIC27", NULL, NULL }, { 0x54584e20, 0xffffffff, "TLC320AD9xC", NULL, NULL }, { 0x56494161, 0xffffffff, "VIA1612A", NULL, NULL }, // modified ICE1232 with S/PDIF { 0x56494170, 0xffffffff, "VIA1617A", patch_vt1617a, NULL }, // modified VT1616 with S/PDIF -- Ondrej Zary

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Re: [alsa-devel] linux-next: Tree for Sep 12 (sound/soc/)
by Randy Dunlap 12 Sep '13

12 Sep '13

On 09/11/13 21:34, Stephen Rothwell wrote: > Hi all, > > Please do not add any code for v3.13 to your linux-next included branches > until after v3.12-rc1 is released. > > Changes since 20130911: > on i386: when CONFIG_REGMAP_I2C=m and CONFIG_SND_SOC=y sound/built-in.o: In function `snd_soc_codec_set_cache_io': (.text+0x2dbb5): undefined reference to `regmap_init_i2c' Full randconfig file is attached. -- ~Randy

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[alsa-devel] Hello
by Michael Jordan 12 Sep '13

12 Sep '13

Hello Gururaja, If I am trying to ask some questions about the alsa system, do I need to copy somebody directly to get a response? I have posted about 4 questions on the group in the last few months, and I have not even gotten a slap on the hand. The reason I am contacting you is you have mentioned working on the TLV320AIC3X which is the chip used on the OMAPL138 board that we have. I am attempting to write a driver for a PCM3168A which we have already constructed a board for, and plugs into the 100 pin audio expander connecter. I have the basic draft of a driver, but everything in the code seems to be fairly strongly coded for the DA850 boards to use the TLV320AIC. I can prevent it loading, and my module will load, but things are not "connecting" internally. I assumed that I would need to make some changes in the arch/arm/mach-davinci/board-da850-evm.c. However what I am running into is chasing me up a trail that eventually has me modifying mach-types.h and creating a whole new board configuration. I said ok, so I copied board-da850-evm.c to board-da850-evm-mod.c which made it necessary for me to begin modifying some Kconfig's, etc., and I still cannot get the kernel to compile with the new kernel driver. I just need to know, am I heading down the wrong path with this? I have a pcm3168a.c and pcm3168a.h which have a basic structure for the new codec, but I cannot figure out how to get the system to actually use it because it basically appears that the TLV320AIC3X is "hard-wired" into the system. Help please, Michael

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[alsa-devel] Fwd: Lynx HiLo USB DAC issues
by Radoslaw Szkodzinski 12 Sep '13

12 Sep '13

Hello, I'm now an owner of a Lynx Hilo USB DAC/ADC. It is supposedly USB Audio Class 2 compatible, but it does require an advanced driver on both Windows and Mac OS X. The device has a complex UAC2 mixer and extra features. It enumerates fine, but there's some problem with the clock selector and sink parameters. I've attached lsusb -vvv and dmesg output. Apparently bNrInPins is invalid for the clock selector for some reason. After adding an extra log of that variable, I get: [49574.136478] __uac_clock_find_source(): selector reported illegal value, id 40, ret 6, pins 1 This can actually be correct if the current clock source is not supported via USB. HiLo has a separate clock source variant for USB. I think that in this case, the clock shouldn't be rejected as invalid, instead switched to the available source if possible. During initialization, there can be any kind of garbage present in the selector. Thanks for any help, -- Radosław Szkodziński

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