October 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: USB-audio: Add support for Novation Nocturn MIDIcontrol surface
by Ricard Wanderlof 18 Oct '15

18 Oct '15

The Nocturn needs the MIDI_RAW_BYTES quirk, like other Novation devices. Tested that the Nocturn shows up in aconnect, and that it can be used as a control surface (using the xtor synthesizer patch editor). Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- I've noticed that checkpatch complains about the space between the & and (, but in the interest of consistency I've not changed this. Is there some reason for the space, or would it be worth while cleaning up the file? sound/usb/quirks-table.h | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index e475665..99de061 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2664,6 +2664,15 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { + USB_DEVICE(0x1235, 0x000a), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "Novation", */ + /* .product_name = "Nocturn", */ + .ifnum = 0, + .type = QUIRK_MIDI_RAW_BYTES + } +}, +{ USB_DEVICE(0x1235, 0x000e), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "Novation", */ -- 2.1.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

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[alsa-devel] [PATCH] ALSA: fireworks/bebob/oxfw/dice: enable to make as built-in
by Takashi Sakamoto 18 Oct '15

18 Oct '15

When committed to upstream, these four modules had wrong entries for Makefile. This forces them to be loadable modules even if they're set as built-in. This commit fixes this bug. Fixes: b5b04336015e('ALSA: fireworks: Add skelton for Fireworks based devices') Fixes: fd6f4b0dc167('ALSA: bebob: Add skelton for BeBoB based devices') Fixes: 1a4e39c2e5ca('ALSA: oxfw: Move to its own directory') Fixes: 14ff6a094815('ALSA: dice: Move file to its own directory') Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/Makefile | 2 +- sound/firewire/dice/Makefile | 2 +- sound/firewire/fireworks/Makefile | 2 +- sound/firewire/oxfw/Makefile | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/firewire/bebob/Makefile b/sound/firewire/bebob/Makefile index 6cf470c..af7ed66 100644 --- a/sound/firewire/bebob/Makefile +++ b/sound/firewire/bebob/Makefile @@ -1,4 +1,4 @@ snd-bebob-objs := bebob_command.o bebob_stream.o bebob_proc.o bebob_midi.o \ bebob_pcm.o bebob_hwdep.o bebob_terratec.o bebob_yamaha.o \ bebob_focusrite.o bebob_maudio.o bebob.o -obj-m += snd-bebob.o +obj-$(CONFIG_SND_BEBOB) += snd-bebob.o diff --git a/sound/firewire/dice/Makefile b/sound/firewire/dice/Makefile index 9ef228e..55b4be9 100644 --- a/sound/firewire/dice/Makefile +++ b/sound/firewire/dice/Makefile @@ -1,3 +1,3 @@ snd-dice-objs := dice-transaction.o dice-stream.o dice-proc.o dice-midi.o \ dice-pcm.o dice-hwdep.o dice.o -obj-m += snd-dice.o +obj-$(CONFIG_SND_DICE) += snd-dice.o diff --git a/sound/firewire/fireworks/Makefile b/sound/firewire/fireworks/Makefile index 0c74408..15ef7f7 100644 --- a/sound/firewire/fireworks/Makefile +++ b/sound/firewire/fireworks/Makefile @@ -1,4 +1,4 @@ snd-fireworks-objs := fireworks_transaction.o fireworks_command.o \ fireworks_stream.o fireworks_proc.o fireworks_midi.o \ fireworks_pcm.o fireworks_hwdep.o fireworks.o -obj-m += snd-fireworks.o +obj-$(CONFIG_SND_FIREWORKS) += snd-fireworks.o diff --git a/sound/firewire/oxfw/Makefile b/sound/firewire/oxfw/Makefile index a926850..06ff50f 100644 --- a/sound/firewire/oxfw/Makefile +++ b/sound/firewire/oxfw/Makefile @@ -1,3 +1,3 @@ snd-oxfw-objs := oxfw-command.o oxfw-stream.o oxfw-control.o oxfw-pcm.o \ oxfw-proc.o oxfw-midi.o oxfw-hwdep.o oxfw.o -obj-m += snd-oxfw.o +obj-$(CONFIG_SND_OXFW) += snd-oxfw.o -- 2.1.4

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[alsa-devel] [PATCH 2/3] ALSA: hda - Remove leftover snd_hda_bus() prototype
by Takashi Iwai 17 Oct '15

17 Oct '15

It was forgotten to be removed. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.h | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 5bcfe7407d83..373fcad840ea 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -301,10 +301,6 @@ struct hda_codec { /* * constructors */ -int snd_hda_bus_new(struct snd_card *card, - const struct hdac_bus_ops *ops, - const struct hdac_io_ops *io_ops, - struct hda_bus **busp); int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, unsigned int codec_addr, struct hda_codec **codecp); int snd_hda_codec_configure(struct hda_codec *codec); -- 2.6.0

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[alsa-devel] [PATCH 1/3] ALSA: hda - Fix bogus codec address check for mixer name assignment
by Takashi Iwai 17 Oct '15

17 Oct '15

The recent commit [7fbe824a0f0e: ALSA: hda - Update mixer name for the lower codec address] tried to improve the mixer chip name assignment in the order of codec address. However, this fix was utterly bogus; it checks the field set in each codec, thus this value is reset at each codec creation, of course. For really handling this priority, the assignment has to be remembered in the common place, namely in hda_bus, instead of hda_codec. Fixes: 7fbe824a0f0e ('ALSA: hda - Update mixer name for the lower codec address') Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_bind.c | 4 ++-- sound/pci/hda/hda_codec.c | 1 - sound/pci/hda/hda_codec.h | 2 +- sound/pci/hda/hda_controller.c | 1 + 4 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index 152acdaa0a45..70671ad65d24 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -63,11 +63,11 @@ int snd_hda_codec_set_name(struct hda_codec *codec, const char *name) /* update the mixer name */ if (!*codec->card->mixername || - codec->mixer_assigned >= codec->core.addr) { + codec->bus->mixer_assigned >= codec->core.addr) { snprintf(codec->card->mixername, sizeof(codec->card->mixername), "%s %s", codec->core.vendor_name, codec->core.chip_name); - codec->mixer_assigned = codec->core.addr; + codec->bus->mixer_assigned = codec->core.addr; } return 0; diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 0e55c6a6cc7e..2eeaf5ea20f9 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -851,7 +851,6 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, INIT_DELAYED_WORK(&codec->jackpoll_work, hda_jackpoll_work); codec->depop_delay = -1; codec->fixup_id = HDA_FIXUP_ID_NOT_SET; - codec->mixer_assigned = -1; #ifdef CONFIG_PM codec->power_jiffies = jiffies; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 055a80522282..5bcfe7407d83 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -70,6 +70,7 @@ struct hda_bus { unsigned int no_response_fallback:1; /* don't fallback at RIRB error */ int primary_dig_out_type; /* primary digital out PCM type */ + unsigned int mixer_assigned; /* codec addr for mixer name */ }; /* from hdac_bus to hda_bus */ @@ -260,7 +261,6 @@ struct hda_codec { unsigned long power_off_acct; unsigned long power_jiffies; #endif - unsigned int mixer_assigned; /* filter the requested power state per nid */ unsigned int (*power_filter)(struct hda_codec *codec, hda_nid_t nid, diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 944455997fdc..d6b93a20361b 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1045,6 +1045,7 @@ int azx_bus_init(struct azx *chip, const char *model, mutex_init(&bus->prepare_mutex); bus->pci = chip->pci; bus->modelname = model; + bus->mixer_assigned = -1; bus->core.snoop = azx_snoop(chip); if (chip->get_position[0] != azx_get_pos_lpib || chip->get_position[1] != azx_get_pos_lpib) -- 2.6.0

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[alsa-devel] [patch 2/2] ALSA: firewire-tascam: off by one in handle_midi_tx()
by Dan Carpenter 17 Oct '15

17 Oct '15

My static checker complains because tscm->spec->midi_capture_ports is either 2 or 4 but the tscm->tx_midi_substreams[] array has 4 elements so this is possibly off by one. I have looked at the code and I think it should be >= instead of > as well. Fixes: 107cc0129a68 ('ALSA: firewire-tascam: add support for incoming MIDI messages by asynchronous transaction') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/firewire/tascam/tascam-transaction.c b/sound/firewire/tascam/tascam-transaction.c index 45c3ce3..d4f64ae 100644 --- a/sound/firewire/tascam/tascam-transaction.c +++ b/sound/firewire/tascam/tascam-transaction.c @@ -158,7 +158,7 @@ static void handle_midi_tx(struct fw_card *card, struct fw_request *request, port = b[0] >> 4; /* TODO: support virtual MIDI ports. */ - if (port > tscm->spec->midi_capture_ports) + if (port >= tscm->spec->midi_capture_ports) goto end; /* Assume the message length. */

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[alsa-devel] [patch 1/2] ALSA: firewire-tascam: fix an LED bug
by Dan Carpenter 17 Oct '15

17 Oct '15

We recently tried to add some new code to support turning the LED on and off but the code in snd_tscm_transaction_reregister() is unreachable. Fixes: e65e2cb99e44 ('ALSA: firewire-tascam: Turn on/off FireWire LED') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/firewire/tascam/tascam-transaction.c b/sound/firewire/tascam/tascam-transaction.c index 1c9a88b..45c3ce3 100644 --- a/sound/firewire/tascam/tascam-transaction.c +++ b/sound/firewire/tascam/tascam-transaction.c @@ -249,7 +249,7 @@ int snd_tscm_transaction_reregister(struct snd_tscm *tscm) /* Turn on messaging. */ reg = cpu_to_be32(0x00000001); - return snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST, + err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST, TSCM_ADDR_BASE + TSCM_OFFSET_MIDI_TX_ON, &reg, sizeof(reg), 0); if (err < 0)

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[alsa-devel] [PATCH] ASoC: rt5645: Recheck the jack detect status after resuming from S3
by Oder Chiou 16 Oct '15

16 Oct '15

The patch rechecks the jack detect status after resuming from S3. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5645.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 89ba4ff..4d54999 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2745,6 +2745,10 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec, snd_soc_update_bits(codec, RT5645_PWR_ANLG1, RT5645_PWR_FV1 | RT5645_PWR_FV2, RT5645_PWR_FV1 | RT5645_PWR_FV2); + if (rt5645->en_button_func && + snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) + queue_delayed_work(system_power_efficient_wq, + &rt5645->jack_detect_work, msecs_to_jiffies(0)); break; case SND_SOC_BIAS_OFF: -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH RESEND] ASoC: rt298: correct index default value
by Bard Liao 16 Oct '15

16 Oct '15

Some of the default value on rt298_index_def are incorrect. Change them to the correct value. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- This patch is base on topic/rt298 branch. --- sound/soc/codecs/rt298.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index ff126a7..8c5cb7e 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -50,24 +50,24 @@ struct rt298_priv { }; static struct reg_default rt298_index_def[] = { - { 0x01, 0xaaaa }, - { 0x02, 0x8aaa }, + { 0x01, 0xa5a8 }, + { 0x02, 0x8e95 }, { 0x03, 0x0002 }, - { 0x04, 0xaf01 }, - { 0x08, 0x000d }, - { 0x09, 0xd810 }, - { 0x0a, 0x0120 }, + { 0x04, 0xaf67 }, + { 0x08, 0x200f }, + { 0x09, 0xd010 }, + { 0x0a, 0x0100 }, { 0x0b, 0x0000 }, { 0x0d, 0x2800 }, - { 0x0f, 0x0000 }, - { 0x19, 0x0a17 }, + { 0x0f, 0x0022 }, + { 0x19, 0x0217 }, { 0x20, 0x0020 }, { 0x33, 0x0208 }, { 0x46, 0x0300 }, - { 0x49, 0x0004 }, - { 0x4f, 0x50e9 }, - { 0x50, 0x2000 }, - { 0x63, 0x2902 }, + { 0x49, 0x4004 }, + { 0x4f, 0x50c9 }, + { 0x50, 0x3000 }, + { 0x63, 0x1b02 }, { 0x67, 0x1111 }, { 0x68, 0x1016 }, { 0x69, 0x273f }, -- 1.8.1.1.439.g50a6b54

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[alsa-devel] Remove Renesas sound SRU for Gen1 support
by Kuninori Morimoto 16 Oct '15

16 Oct '15

Hi ALSA, and SH ML I would like to remove Renesas sound SRU support of Gen1 in the future. It was created for preparation of Gen2 SRC support, and it converts sampling rate. But no-one is using this feature in Gen1. board-bockw had used SRU before, but it was pass through mode. This means it is same as SSI. 2nd reason is that board-bockw was removed from upstream code. BockW is now supported by DT, but it doesn't use SRU. More detail, r8a7778.dtsi has rcar_sound,src entry, but no-one is using this feature. SRU proving has no relation to this removing feature. This means there is no effect for DT compatibility. Gen2 SRC was created from Gen1 SRU, these are similar but not same IP. Keeping Gen1 SRU in current driver is a little bit difficult, and no-one is using it today. Gen1 sound is still supported via SSI. If there is no objection, I would like to remove it from upstream step1. add warning message in v4.x step2. remove it in v4.x+1 Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH] ASoC: rsnd: Gen1 probe is not error
by Kuninori Morimoto 16 Oct '15

16 Oct '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Probing from Gen1 is not error. This patch fixup it Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/ctu.c | 6 ++---- sound/soc/sh/rcar/dvc.c | 6 ++---- sound/soc/sh/rcar/mix.c | 6 ++---- 3 files changed, 6 insertions(+), 12 deletions(-) diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c index 6e34909c..27d56a0 100644 --- a/sound/soc/sh/rcar/ctu.c +++ b/sound/soc/sh/rcar/ctu.c @@ -125,10 +125,8 @@ int rsnd_ctu_probe(struct platform_device *pdev, int i, nr, ret; /* This driver doesn't support Gen1 at this point */ - if (rsnd_is_gen1(priv)) { - dev_warn(dev, "CTU is not supported on Gen1\n"); - return -EINVAL; - } + if (rsnd_is_gen1(priv)) + return 0; rsnd_of_parse_ctu(pdev, of_data, priv); diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index 08e58d0..63e8e52 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -338,10 +338,8 @@ int rsnd_dvc_probe(struct platform_device *pdev, int i, nr, ret; /* This driver doesn't support Gen1 at this point */ - if (rsnd_is_gen1(priv)) { - dev_warn(dev, "CMD is not supported on Gen1\n"); - return -EINVAL; - } + if (rsnd_is_gen1(priv)) + return 0; rsnd_of_parse_dvc(pdev, of_data, priv); diff --git a/sound/soc/sh/rcar/mix.c b/sound/soc/sh/rcar/mix.c index 2311d49..a5485b2 100644 --- a/sound/soc/sh/rcar/mix.c +++ b/sound/soc/sh/rcar/mix.c @@ -158,10 +158,8 @@ int rsnd_mix_probe(struct platform_device *pdev, int i, nr, ret; /* This driver doesn't support Gen1 at this point */ - if (rsnd_is_gen1(priv)) { - dev_warn(dev, "MIX is not supported on Gen1\n"); - return -EINVAL; - } + if (rsnd_is_gen1(priv)) + return 0; rsnd_of_parse_mix(pdev, of_data, priv); -- 1.9.1

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