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

[alsa-devel] [PATCH 0/5 v4] ALSA: dice: improve card registration processing
by Takashi Sakamoto 30 Dec '15

30 Dec '15

Hi, This patchset updates my previous one: [alsa-devel] [PATCH 0/5 v3] ALSA: dice: improve card registration processing http://mailman.alsa-project.org/pipermail/alsa-devel/2015-December/102501.h… The patch 02 included a bug of missing 'end' tag. This v4 patchset includes fixes of the bug. Additionally, this patchset adds to output information message when registration work encounters any errors. Furthermore, I use mod_delayed_work() to queue and postpone work for sound card registration. I've reported system freeze in previous patchset, but I had a mistake not to use any actual workqueus as a first argument of the function. Takashi Sakamoto (5): ALSA: dice: split subaddress check from category check ALSA: dice: postpone card registration ALSA: dice: ensure to release sound devices after sound card registration fails ALSA: dice: purge transaction initialization at timeout of Dice notification ALSA: dice: expand timeout to wait for Dice notification sound/firewire/dice/dice-midi.c | 2 + sound/firewire/dice/dice-pcm.c | 2 + sound/firewire/dice/dice-transaction.c | 122 ++++++++++++-------- sound/firewire/dice/dice.c | 197 +++++++++++++++++---------------- sound/firewire/dice/dice.h | 5 + 5 files changed, 189 insertions(+), 139 deletions(-) -- 2.5.0

2 12

[alsa-devel] Rawmidi
by Daniel Viksporre 29 Dec '15

29 Dec '15

I cant compile example programs that use rawmidi. When I look on my harddrive, I have many rawmidi.h files, but since 2015 they are all 0 bytes. Can that be the reason I cant compile. But why is those files 0 bytes from the repository Im using? I get the following error messages... test.c:(.text+0x507): undefined reference to `snd_rawmidi_open' test.c:(.text+0x5d8): undefined reference to `snd_rawmidi_open' test.c:(.text+0x75e): undefined reference to `snd_rawmidi_read' test.c:(.text+0x846): undefined reference to `snd_rawmidi_write' test.c:(.text+0x867): undefined reference to `snd_rawmidi_write' test.c:(.text+0x888): undefined reference to `snd_rawmidi_write' test.c:(.text+0x894): undefined reference to `snd_rawmidi_drain' test.c:(.text+0x8c1): undefined reference to `snd_rawmidi_write' test.c:(.text+0x8e2): undefined reference to `snd_rawmidi_write' test.c:(.text+0x903): undefined reference to `snd_rawmidi_write' test.c:(.text+0x90f): undefined reference to `snd_rawmidi_drain' test.c:(.text+0xa71): undefined reference to `snd_rawmidi_read' test.c:(.text+0xae0): undefined reference to `snd_rawmidi_write' test.c:(.text+0xaec): undefined reference to `snd_rawmidi_drain' test.c:(.text+0xb8f): undefined reference to `snd_rawmidi_drain' test.c:(.text+0xb9b): undefined reference to `snd_rawmidi_close' test.c:(.text+0xbaf): undefined reference to `snd_rawmidi_drain' test.c:(.text+0xbbb): undefined reference to `snd_rawmidi_close'

4 6

[alsa-devel] [PATCH] ASoC: arizona: Exit startup early if no runtime
by Charles Keepax 29 Dec '15

29 Dec '15

commit 9b8ef9f6b3fc ("ASoC: dapm: Add startup & shutdown for dai_links") Added support for calling startup on CODEC to CODEC links, however this is called with a NULL runtime pointer. There isn't really a sensible way to pass a valid runtime pointer to a CODEC to CODEC link at the moment, so we need to make the startup function safe for NULL runtimes. This patch returns from the Arizona startup function early if there is no runtime, this is perfectly safe as all the startup function does is set the PCM constraints for user-space which arn't relevant to a CODEC to CODEC link anyway. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index fb8f188..33143fe 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1494,6 +1494,9 @@ static int arizona_startup(struct snd_pcm_substream *substream, const struct snd_pcm_hw_constraint_list *constraint; unsigned int base_rate; + if (!substream->runtime) + return 0; + switch (dai_priv->clk) { case ARIZONA_CLK_SYSCLK: base_rate = priv->sysclk; -- 2.1.4

1 0

[alsa-devel] [PATCH v2 1/2] ALSA: hda - Add mic mute hotkey quirk for Lenovo ThinkCentre AIO
by Hui Wang 29 Dec '15

29 Dec '15

From: Kailang <kailang(a)realtek.com> The Lenovo ThinkCenter AIO uses Line2 (NID 0x1b) to implement the micmute hotkey, here we register an input device and use Line2 unsol event to collect the hotkey pressing or releasing. In the meanwhile, the micmute led is controlled by GPIO2, so we use an existing function alc_fixup_gpio_mic_mute_hook() to control the led. [Hui: And there are two places to register the input device, to make the code simple and clean, move the two same code sections into a function.] Cc: <stable(a)vger.kernel.org> Signed-off-by: Kailang <kailang(a)realtek.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 84 ++++++++++++++++++++++++++++++++++++------- 1 file changed, 71 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fe96428..257b839 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3468,6 +3468,29 @@ static void gpio2_mic_hotkey_event(struct hda_codec *codec, input_sync(spec->kb_dev); } +static int alc_register_micmute_input_device(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->kb_dev = input_allocate_device(); + if (!spec->kb_dev) { + codec_err(codec, "Out of memory (input_allocate_device)\n"); + return -ENOMEM; + } + spec->kb_dev->name = "Microphone Mute Button"; + spec->kb_dev->evbit[0] = BIT_MASK(EV_KEY); + spec->kb_dev->keybit[BIT_WORD(KEY_MICMUTE)] = BIT_MASK(KEY_MICMUTE); + + if (input_register_device(spec->kb_dev)) { + codec_err(codec, "input_register_device failed\n"); + input_free_device(spec->kb_dev); + spec->kb_dev = NULL; + return -ENOMEM; + } + + return 0; +} + static void alc280_fixup_hp_gpio2_mic_hotkey(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -3485,20 +3508,8 @@ static void alc280_fixup_hp_gpio2_mic_hotkey(struct hda_codec *codec, struct alc_spec *spec = codec->spec; if (action == HDA_FIXUP_ACT_PRE_PROBE) { - spec->kb_dev = input_allocate_device(); - if (!spec->kb_dev) { - codec_err(codec, "Out of memory (input_allocate_device)\n"); + if (alc_register_micmute_input_device(codec) != 0) return; - } - spec->kb_dev->name = "Microphone Mute Button"; - spec->kb_dev->evbit[0] = BIT_MASK(EV_KEY); - spec->kb_dev->keybit[BIT_WORD(KEY_MICMUTE)] = BIT_MASK(KEY_MICMUTE); - if (input_register_device(spec->kb_dev)) { - codec_err(codec, "input_register_device failed\n"); - input_free_device(spec->kb_dev); - spec->kb_dev = NULL; - return; - } snd_hda_add_verbs(codec, gpio_init); snd_hda_codec_write_cache(codec, codec->core.afg, 0, @@ -3528,6 +3539,47 @@ static void alc280_fixup_hp_gpio2_mic_hotkey(struct hda_codec *codec, } } +static void alc233_fixup_lenovo_line2_mic_hotkey(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + /* Line2 = mic mute hotkey + GPIO2 = mic mute LED */ + static const struct hda_verb gpio_init[] = { + { 0x01, AC_VERB_SET_GPIO_MASK, 0x04 }, + { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x04 }, + {} + }; + + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + if (alc_register_micmute_input_device(codec) != 0) + return; + + snd_hda_add_verbs(codec, gpio_init); + snd_hda_jack_detect_enable_callback(codec, 0x1b, + gpio2_mic_hotkey_event); + + spec->gen.cap_sync_hook = alc_fixup_gpio_mic_mute_hook; + spec->gpio_led = 0; + spec->mute_led_polarity = 0; + spec->gpio_mic_led_mask = 0x04; + return; + } + + if (!spec->kb_dev) + return; + + switch (action) { + case HDA_FIXUP_ACT_PROBE: + spec->init_amp = ALC_INIT_DEFAULT; + break; + case HDA_FIXUP_ACT_FREE: + input_unregister_device(spec->kb_dev); + spec->kb_dev = NULL; + } +} + static void alc269_fixup_hp_line1_mic1_led(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -4628,6 +4680,7 @@ enum { ALC275_FIXUP_DELL_XPS, ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE, ALC293_FIXUP_LENOVO_SPK_NOISE, + ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, }; static const struct hda_fixup alc269_fixups[] = { @@ -5237,6 +5290,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_THINKPAD_ACPI }, + [ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc233_fixup_lenovo_line2_mic_hotkey, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -5386,6 +5443,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2223, "ThinkPad T550", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE), + SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- 1.9.1

2 2

[alsa-devel] [PATCH 1/2] ALSA: hda - Add mic mute hotkey quirk for Lenovo ThinkCentre AIO
by Hui Wang 28 Dec '15

28 Dec '15

From: Kailang <kailang(a)realtek.com> The Lenovo ThinkCenter AIO uses Line2 (NID 0x1b) to implement the micmute hotkey, here we register an input device and use Line2 unsol event to collect the hotkey pressing or releasing. In the meanwhile, the micmute led is controlled by GPIO2, so we use an existing function alc_fixup_gpio_mic_mute_hook() to control the led. Cc: <stable(a)vger.kernel.org> Signed-off-by: Kailang <kailang(a)realtek.com> --- sound/pci/hda/patch_realtek.c | 59 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 59 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fe96428..8adff4f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3528,6 +3528,59 @@ static void alc280_fixup_hp_gpio2_mic_hotkey(struct hda_codec *codec, } } +static void alc233_fixup_lenovo_line2_mic_hotkey(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + /* Line2 = mic mute hotkey + GPIO2 = mic mute LED */ + static const struct hda_verb gpio_init[] = { + { 0x01, AC_VERB_SET_GPIO_MASK, 0x04 }, + { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x04 }, + {} + }; + + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->kb_dev = input_allocate_device(); + if (!spec->kb_dev) { + codec_err(codec, "Out of memory (input_allocate_device)\n"); + return; + } + spec->kb_dev->name = "Microphone Mute Button"; + spec->kb_dev->evbit[0] = BIT_MASK(EV_KEY); + spec->kb_dev->keybit[BIT_WORD(KEY_MICMUTE)] = BIT_MASK(KEY_MICMUTE); + if (input_register_device(spec->kb_dev)) { + codec_err(codec, "input_register_device failed\n"); + input_free_device(spec->kb_dev); + spec->kb_dev = NULL; + return; + } + + snd_hda_add_verbs(codec, gpio_init); + snd_hda_jack_detect_enable_callback(codec, 0x1b, + gpio2_mic_hotkey_event); + + spec->gen.cap_sync_hook = alc_fixup_gpio_mic_mute_hook; + spec->gpio_led = 0; + spec->mute_led_polarity = 0; + spec->gpio_mic_led_mask = 0x04; + return; + } + + if (!spec->kb_dev) + return; + + switch (action) { + case HDA_FIXUP_ACT_PROBE: + spec->init_amp = ALC_INIT_DEFAULT; + break; + case HDA_FIXUP_ACT_FREE: + input_unregister_device(spec->kb_dev); + spec->kb_dev = NULL; + } +} + static void alc269_fixup_hp_line1_mic1_led(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -4628,6 +4681,7 @@ enum { ALC275_FIXUP_DELL_XPS, ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE, ALC293_FIXUP_LENOVO_SPK_NOISE, + ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, }; static const struct hda_fixup alc269_fixups[] = { @@ -5237,6 +5291,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_THINKPAD_ACPI }, + [ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc233_fixup_lenovo_line2_mic_hotkey, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -5386,6 +5444,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2223, "ThinkPad T550", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE), + SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- 1.9.1

2 8

[alsa-devel] [PATCH] ASoC: sun4i-codec: Use proper output for external amp routes
by Hans de Goede 27 Dec '15

27 Dec '15

An external amp (if any) is connected to the external outputs of the SoC of course, rather then directly to the internal amp. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/sunxi/sun4i-codec.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index e6cc6a1..44f170c 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -728,7 +728,8 @@ static const struct snd_soc_dapm_widget sun4i_codec_card_dapm_widgets[] = { }; static const struct snd_soc_dapm_route sun4i_codec_card_dapm_routes[] = { - { "Speaker", NULL, "Power Amplifier" }, + { "Speaker", NULL, "HP Right" }, + { "Speaker", NULL, "HP Left" }, }; static struct snd_soc_card *sun4i_codec_create_card(struct device *dev) -- 2.5.0

3 2

[alsa-devel] [PATCH 0/5 v3] ALSA: dice: improve card registration processing
by Takashi Sakamoto 25 Dec '15

25 Dec '15

Hi, This patchset updates my previous post. [alsa-devel] [PATCH 0/4 v2] ALSA: dice: improve card registration processing http://mailman.alsa-project.org/pipermail/alsa-devel/2015-December/102460.h… The previous patchset includes some bugs. * Mutex structure can be destroyed within critical section with it in releasing sound card. * Bus reset handler doesn't execute transaction re-initialization as long as sound card instance is not registered. * Work of sound card registration is not postpone even if bus reset occurs. * Devices in sound card are not released when sound card registration fails. This version fixes these bugs, according to discussions between Takash Iwai and Stefan Richter. Thanks for them. Changes: * Use delayed workqueue API correctly. * Wait for registration work in .remove callback, instead of sound card free callback. * Obsolete mutex_lock()/mutex_unlock() for sound card registration purpose. * Keep devices of sound card and release them at registration failure. In kerneldoc of 'mod_delayed_work', this function call has a side effect to queueing the work. It seems that this function can be used instead of schedule_delayed_work. https://www.kernel.org/doc/htmldocs/device-drivers/API-mod-delayed-work.html But as long as I tested, my Linux 4.2 system is freezed. I have no further investigation. I use both of schedule_delayed_work() and mod_delayed_work() in this patchset. Takashi Sakamoto (5): ALSA: dice: split subaddress check from category check ALSA: dice: postpone card registration ALSA: dice: ensure to release sound devices after sound card registration fails ALSA: dice: purge transaction initialization at timeout of Dice notification ALSA: dice: expand timeout to wait for Dice notification sound/firewire/dice/dice-midi.c | 2 + sound/firewire/dice/dice-pcm.c | 2 + sound/firewire/dice/dice-transaction.c | 122 ++++++++++++-------- sound/firewire/dice/dice.c | 199 +++++++++++++++++---------------- sound/firewire/dice/dice.h | 5 + 5 files changed, 191 insertions(+), 139 deletions(-) -- 2.5.0

1 5

[alsa-devel] [PATCH 0/4 v2] ALSA: dice: improve card registration processing
by Takashi Sakamoto 25 Dec '15

25 Dec '15

Hi, This patchset is to update my previous one: [alsa-devel] [PATCH 0/4] ALSA: dice: improve card registration processing http://mailman.alsa-project.org/pipermail/alsa-devel/2015-December/102357.h… The main purpose of this patchset is to postpone card registration after successive bus resets, to ensure the processing and related transactions. This change has an advantage for ALSA dice driver to get proper information about hardware, especially for Dice II based models. Changes from v1: * fix jiffies64 handling bug * rename some members and functions related to card registration. * 'struct snd_dice.probed' -> 'registered' * do_probe() -> do_registration() Takashi Sakamoto (4): ALSA: dice: split subaddress check from category check ALSA: dice: postpone card registration ALSA: dice: purge transaction initialization at timeout of Dice notification ALSA: dice: expand timeout to wait for Dice notification sound/firewire/dice/dice-transaction.c | 122 ++++++++++++++-------- sound/firewire/dice/dice.c | 183 ++++++++++++++++----------------- sound/firewire/dice/dice.h | 3 + 3 files changed, 172 insertions(+), 136 deletions(-) -- 2.5.0

2 11

[alsa-devel] [RFC v2][PATCH 00/11] ALSA: fireface: new driver for RME Fireface series
by Takashi Sakamoto 24 Dec '15

24 Dec '15

Hi, This patchset is to update my previous one with mostly full features as an driver in ALSA firewire stack: [alsa-devel] [RFC][PATCH 0/3] ALSA: fireface: new driver for RME Fireface series (MIDI only) http://mailman.alsa-project.org/pipermail/alsa-devel/2015-December/101576.h… Changes from RFCv1: * Support own quirk of packet streaming * Add own data block processing layer * Support PCM functionality * Add proc node to help debugging * Add hwdep interface I note that just powering on, the device is muted. Thus, the device generates no sound even if packet streaming is successful. You can de-mute it by write transaction. For example, by using 'firewire-request' command in linux-firewire-utils(https://github.com/cladisch/linux-firewire-utils): $ ./firewire-request /dev/fw1 write 0x0000801c0000 0x000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000001000000010000000100000001000000 Currently, libffado gives no way to do it via its mixer interface. Instead, done in methods to start streaming. I think it better to move de-mute codes to mixer initialization. There's an issue of periodical hissing noise. I observed this with Fireface 400. Interval of the noise differs depending on the situation (10-20 minutes) and continues around 20 seconds. The cause is not clear but I've experienced similar issue with TASCAM FireOne. A common point between these two models is non-blocking transmission. I guess that the cause may be on data block calculation in AMDTP packet streaming layer (calculate_data_blocks() function in sound/firewire/amdtp-stream.c). Further investigation is required. In my previous RFC, I described that zero bits have no effect in write transactions to 0x00008010051c. But this is wrong. The transaction has side effect to set the other options. In this patchset, the state of clock is changed on the way to initialize transaction. This is a bit rude but unavoidable. Fortunately, it's software's responsibility to read from flash memory and set initial configuration, thus I think this change is approval because just powering on the state of device is against user's configuration. The loading and configuring are still userspace responsibility. MIDI functionality can be disabled when any userspace applications run with libffado, as I described. This is a bug of libffado due to node ID handling. In line 199 of libffado/src/rme/fireface_hw.cpp, node id is handled as uint16_t value. This is a fashion of libffado to represent Node ID by lower 6 bits of actual Node ID of IEEE 1394 bus. In the other lines, actual Node Id is constructed with 0xffc0. Just using the raw value returned by ConfigRom::getNodeId() is invalid. For this work, I can borrow test device (Fireface 400) from Syntax Japan Inc. It's my fortune to have an opportunity to contact to the company. I'm graceful to Takeshi Mitsuhashi, Tahiro Hashizume and their kindness. Thank you. Takashi Sakamoto (11): fireface: add skeleton for RME Fireface series fireface: postpone sound card registration fireface: add model specific structure fireface: add transaction support fireface: add support for MIDI functionality fireface: add proc node to help debugging firewire-lib: add no-header packet processing fireface: add data processing layer fireface: add stream management functionality fireface: add PCM functionality fireface: add hwdep interface include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 1 + sound/firewire/Kconfig | 7 + sound/firewire/Makefile | 1 + sound/firewire/amdtp-stream.c | 104 +++++-- sound/firewire/amdtp-stream.h | 2 + sound/firewire/fireface/Makefile | 4 + sound/firewire/fireface/amdtp-ff.c | 221 ++++++++++++++ sound/firewire/fireface/fireface-hwdep.c | 191 ++++++++++++ sound/firewire/fireface/fireface-midi.c | 131 ++++++++ sound/firewire/fireface/fireface-pcm.c | 390 ++++++++++++++++++++++++ sound/firewire/fireface/fireface-proc.c | 237 +++++++++++++++ sound/firewire/fireface/fireface-stream.c | 397 +++++++++++++++++++++++++ sound/firewire/fireface/fireface-transaction.c | 301 +++++++++++++++++++ sound/firewire/fireface/fireface.c | 210 +++++++++++++ sound/firewire/fireface/fireface.h | 131 ++++++++ 16 files changed, 2313 insertions(+), 18 deletions(-) create mode 100644 sound/firewire/fireface/Makefile create mode 100644 sound/firewire/fireface/amdtp-ff.c create mode 100644 sound/firewire/fireface/fireface-hwdep.c create mode 100644 sound/firewire/fireface/fireface-midi.c create mode 100644 sound/firewire/fireface/fireface-pcm.c create mode 100644 sound/firewire/fireface/fireface-proc.c create mode 100644 sound/firewire/fireface/fireface-stream.c create mode 100644 sound/firewire/fireface/fireface-transaction.c create mode 100644 sound/firewire/fireface/fireface.c create mode 100644 sound/firewire/fireface/fireface.h -- 2.5.0

3 18

[alsa-devel] [PATCH] ASoC: wm8960: add kcontrol to select ADC data output
by Zidan Wang 24 Dec '15

24 Dec '15

add kcontrol to select ADC data output. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/codecs/wm8960.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 5380798..162def8 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -147,6 +147,12 @@ static const char *wm8960_3d_upper_cutoff[] = {"High", "Low"}; static const char *wm8960_3d_lower_cutoff[] = {"Low", "High"}; static const char *wm8960_alcfunc[] = {"Off", "Right", "Left", "Stereo"}; static const char *wm8960_alcmode[] = {"ALC", "Limiter"}; +static const char *wm8960_adc_data_output_sel[] = { + "Left Data = Left ADC; Right Data = Right ADC", + "Left Data = Left ADC; Right Data = Left ADC", + "Left Data = Right ADC; Right Data = Right ADC", + "Left Data = Right ADC; Right Data = Left ADC", +}; static const struct soc_enum wm8960_enum[] = { SOC_ENUM_SINGLE(WM8960_DACCTL1, 5, 4, wm8960_polarity), @@ -155,6 +161,7 @@ static const struct soc_enum wm8960_enum[] = { SOC_ENUM_SINGLE(WM8960_3D, 5, 2, wm8960_3d_lower_cutoff), SOC_ENUM_SINGLE(WM8960_ALC1, 7, 4, wm8960_alcfunc), SOC_ENUM_SINGLE(WM8960_ALC3, 8, 2, wm8960_alcmode), + SOC_ENUM_SINGLE(WM8960_ADDCTL1, 2, 4, wm8960_adc_data_output_sel), }; static const int deemph_settings[] = { 0, 32000, 44100, 48000 }; @@ -295,6 +302,8 @@ SOC_SINGLE_TLV("Right Output Mixer Boost Bypass Volume", WM8960_BYPASS2, 4, 7, 1, bypass_tlv), SOC_SINGLE_TLV("Right Output Mixer RINPUT3 Volume", WM8960_ROUTMIX, 4, 7, 1, bypass_tlv), + +SOC_ENUM("ADC Data Output Select", wm8960_enum[6]), }; static const struct snd_kcontrol_new wm8960_lin_boost[] = { -- 1.9.1

2 1