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

[alsa-devel] [PATCH 1/3] emu10k1: remove unused AC'97 mixer controls on Audigy
by Maciej S. Szmigiero 08 Jul '15

08 Jul '15

AC'97 Headphone output and EAPD control aren't used on Audigy so remove them from mixer. Also remove AC'97 3D control as the driver is already doing for Audigys with 1361T ADC. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/pci/emu10k1/emumixer.c | 11 ++++++----- 1 files changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index 55e5716..f1cb7b3 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -1830,10 +1830,16 @@ int snd_emu10k1_mixer(struct snd_emu10k1 *emu, "Capture Switch", "Capture Volume", "Mic Select", + "Headphone Playback Switch", + "Headphone Playback Volume", + "3D Control - Center", + "3D Control - Depth", + "3D Control - Switch", "Video Playback Switch", "Video Playback Volume", "Mic Playback Switch", "Mic Playback Volume", + "External Amplifier", NULL }; static char *audigy_rename_ctls[] = { @@ -1996,11 +2002,6 @@ int snd_emu10k1_mixer(struct snd_emu10k1 *emu, rename_ctl(card, "Analog Mix Capture Volume", "Line2 Capture Volume"); rename_ctl(card, "Aux2 Capture Volume", "Line3 Capture Volume"); rename_ctl(card, "Mic Capture Volume", "Unknown1 Capture Volume"); - remove_ctl(card, "Headphone Playback Switch"); - remove_ctl(card, "Headphone Playback Volume"); - remove_ctl(card, "3D Control - Center"); - remove_ctl(card, "3D Control - Depth"); - remove_ctl(card, "3D Control - Switch"); } if ((kctl = emu->ctl_send_routing = snd_ctl_new1(&snd_emu10k1_send_routing_control, emu)) == NULL) return -ENOMEM;

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Re: [alsa-devel] [pulseaudio-discuss] Strange hardware: XFX nForce 780i SLI MCP
by Raymond Yau 08 Jul '15

08 Jul '15

> > Yesterday I got an interesting email related to dcaenc support. > > The person has an "XFX nForce 780i SLI" MCP board: http://www.nvidia.com/object/product_nforce_780i_sli_us.html , http://www.evga.com/support/manuals/files/132-CK-NF78.pdf > > As you see on page 18 of the manual, this board has no HDMI output, but there is an optical SPDIF. Strangely enough, this SPDIF is his "hdmi:CARD=NVidia,DEV=0" device. > > Given the recent discussions related to dynamically creating PCM devices for HDMI, I think you will be interested in this alsa-info output. > > Is there anything here that would need special treatment by ALSA or PulseAudio? Or, is this just a misnumbered/misnamed codec? > Check whether BIOS set the internal spdif as HDMI as pin default The alsa driver prefer HDMI instead of SPDIF when there are two Digital Outs

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[alsa-devel] [PATCH] ASoC: ts3a227e: do not report jack status when there is i2c read err
by yang.a.fang＠intel.com 07 Jul '15

07 Jul '15

From: "Fang, Yang A" <yang.a.fang(a)intel.com> After suspend -> resume the ts3a227e_interrupt sometimes comes before i2c controller resume is called .regmap_read will return incorrect status and report a wrong jack status.We should return if there is read err,the interrupt will come again since it is level triggered and we are not yet clear the interrupt. In addtion,cht_bsw_max98090_ti machine driver registered additional notifier base on jack event which will program the audio codec.there will be codec timeout err if such event occurs prior to i2c controller is resumed. Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> --- sound/soc/codecs/ts3a227e.c | 18 ++++++++++++++++-- 1 file changed, 16 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ts3a227e.c b/sound/soc/codecs/ts3a227e.c index 12232d7..12d0f2a 100644 --- a/sound/soc/codecs/ts3a227e.c +++ b/sound/soc/codecs/ts3a227e.c @@ -23,6 +23,7 @@ #include "ts3a227e.h" struct ts3a227e { + struct device *dev; struct regmap *regmap; struct snd_soc_jack *jack; bool plugged; @@ -189,16 +190,28 @@ static irqreturn_t ts3a227e_interrupt(int irq, void *data) struct ts3a227e *ts3a227e = (struct ts3a227e *)data; struct regmap *regmap = ts3a227e->regmap; unsigned int int_reg, kp_int_reg, acc_reg, i; + struct device *dev = ts3a227e->dev; + int ret; /* Check for plug/unplug. */ - regmap_read(regmap, TS3A227E_REG_INTERRUPT, &int_reg); + ret = regmap_read(regmap, TS3A227E_REG_INTERRUPT, &int_reg); + if (ret) { + dev_err(dev, "failed to clear interrupt ret=%d\n", ret); + return IRQ_HANDLED; + } + if (int_reg & (DETECTION_COMPLETE_EVENT | INS_REM_EVENT)) { regmap_read(regmap, TS3A227E_REG_ACCESSORY_STATUS, &acc_reg); ts3a227e_new_jack_state(ts3a227e, acc_reg); } /* Report any key events. */ - regmap_read(regmap, TS3A227E_REG_KP_INTERRUPT, &kp_int_reg); + ret = regmap_read(regmap, TS3A227E_REG_KP_INTERRUPT, &kp_int_reg); + if (ret) { + dev_err(dev, "failed to clear key interrupt ret=%d\n", ret); + return IRQ_HANDLED; + } + for (i = 0; i < TS3A227E_NUM_BUTTONS; i++) { if (kp_int_reg & PRESS_MASK(i)) ts3a227e->buttons_held |= (1 << i); @@ -283,6 +296,7 @@ static int ts3a227e_i2c_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, ts3a227e); + ts3a227e->dev = dev; ts3a227e->regmap = devm_regmap_init_i2c(i2c, &ts3a227e_regmap_config); if (IS_ERR(ts3a227e->regmap)) -- 1.7.9.5

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[alsa-devel] [PATCH 3/3] emu10k1: rename Audigy Analog Capture Boost control
by Maciej S. Szmigiero 07 Jul '15

07 Jul '15

Audigy has "Analog Capture Boost" mixer control, however now this only controls mic level, not other analog sources. It applies also both to playback and capture, so rename it to something more descriptive. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/pci/emu10k1/emumixer.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index 567b5cb..076b117 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -1741,7 +1741,7 @@ static int snd_audigy_capture_boost_put(struct snd_kcontrol *kcontrol, static struct snd_kcontrol_new snd_audigy_capture_boost = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Analog Capture Boost", + .name = "Mic Extra Boost", .info = snd_audigy_capture_boost_info, .get = snd_audigy_capture_boost_get, .put = snd_audigy_capture_boost_put

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[alsa-devel] [PATCH 2/3] emu10k1: enable TAD mic out on Audigy
by Maciej S. Szmigiero 07 Jul '15

07 Jul '15

Enable TAD output on Audigy naming it "Phone Output", to be consistent with TAD input which is called "Phone". According to Creative doc ( http://support.creative.com/kb/ShowArticle.aspx?sid=3026 ) this should output just mic signal. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/pci/emu10k1/emumixer.c | 12 +++++++----- 1 files changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index f1cb7b3..567b5cb 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -1819,8 +1819,6 @@ int snd_emu10k1_mixer(struct snd_emu10k1 *emu, * the Philips ADC for 24bit capture */ "PCM Playback Switch", "PCM Playback Volume", - "Master Mono Playback Switch", - "Master Mono Playback Volume", "Master Playback Switch", "Master Playback Volume", "PCM Out Path & Mute", @@ -1848,6 +1846,8 @@ int snd_emu10k1_mixer(struct snd_emu10k1 *emu, /* "Wave Capture Volume", "PCM Capture Volume", */ "Wave Master Playback Volume", "Master Playback Volume", "AMic Playback Volume", "Mic Playback Volume", + "Master Mono Playback Switch", "Phone Output Playback Switch", + "Master Mono Playback Volume", "Phone Output Playback Volume", NULL }; static char *audigy_rename_ctls_i2c_adc[] = { @@ -1873,8 +1873,6 @@ int snd_emu10k1_mixer(struct snd_emu10k1 *emu, * the Philips ADC for 24bit capture */ "PCM Playback Switch", "PCM Playback Volume", - "Master Mono Playback Switch", - "Master Mono Playback Volume", "Capture Source", "Capture Switch", "Capture Volume", @@ -1906,7 +1904,8 @@ int snd_emu10k1_mixer(struct snd_emu10k1 *emu, "Aux Playback Volume", "Aux Capture Volume", "Video Playback Switch", "Video Capture Switch", "Video Playback Volume", "Video Capture Volume", - + "Master Mono Playback Switch", "Phone Output Playback Switch", + "Master Mono Playback Volume", "Phone Output Playback Volume", NULL }; @@ -1941,6 +1940,9 @@ int snd_emu10k1_mixer(struct snd_emu10k1 *emu, snd_ac97_write_cache(emu->ac97, AC97_MASTER, 0x0000); /* set capture source to mic */ snd_ac97_write_cache(emu->ac97, AC97_REC_SEL, 0x0000); + /* set mono output (TAD) to mic */ + snd_ac97_update_bits(emu->ac97, AC97_GENERAL_PURPOSE, + 0x0200, 0x0200); if (emu->card_capabilities->adc_1361t) c = audigy_remove_ctls_1361t_adc; else

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[alsa-devel] [PATCH 1/4] ASoC: max98090: Check for clk_prepare_enable() error
by Fabio Estevam 07 Jul '15

07 Jul '15

From: Fabio Estevam <fabio.estevam(a)freescale.com> clk_prepare_enable() may fail, so we should better check its return value and propagate it in the case of error. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/max98090.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 78268f05..1697340 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1801,10 +1801,13 @@ static int max98090_set_bias_level(struct snd_soc_codec *codec, if (IS_ERR(max98090->mclk)) break; - if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_ON) + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_ON) { clk_disable_unprepare(max98090->mclk); - else - clk_prepare_enable(max98090->mclk); + } else { + ret = clk_prepare_enable(max98090->mclk); + if (ret) + return ret; + } break; case SND_SOC_BIAS_STANDBY: -- 1.9.1

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[alsa-devel] [PATCH V2] ASoC: core: Don't probe the component which is dummy
by Shengjiu Wang 07 Jul '15

07 Jul '15

Dummy dai can be used by multiple sound card. But it only belong to one card's dapm list. If another card use it, there will be dapm_assert_locked warning. [ 20.015782] WARNING: CPU: 1 PID: 661 at sound/soc/soc-dapm.c:124 dapm_assert_locked.isra.36+0x4c/0x58() [ 20.025249] Modules linked in: [ 20.028349] CPU: 1 PID: 661 Comm: aplay Not tainted 4.1.0-rc6-next-20150605-00004-gaee05d8-dirty #92 [ 20.037528] Hardware name: Freescale i.MX6 Quad/DualLite (Device Tree) [ 20.044110] Backtrace: [ 20.046614] [<80012e00>] (dump_backtrace) from [<80012fa0>] (show_stack+0x18/0x1c) [ 20.054229] r6:809e8060 r5:00000000 r4:00000000 r3:00000000 [ 20.060002] [<80012f88>] (show_stack) from [<807a0f74>] (dump_stack+0x80/0x9c) [ 20.067293] [<807a0ef4>] (dump_stack) from [<8002b144>] (warn_slowpath_common+0x7c/0xb4) [ 20.075427] r5:0000007c r4:00000000 [ 20.079065] [<8002b0c8>] (warn_slowpath_common) from [<8002b1a0>] (warn_slowpath_null+0x24/0x2c) [ 20.087898] r8:00000001 r7:88007c28 r6:ed94a680 r5:809e83e4 r4:ed83d6c0 [ 20.094747] [<8002b17c>] (warn_slowpath_null) from [<8058403c>] (dapm_assert_locked.isra.36+0x4c/0x58) [ 20.104101] [<80583ff0>] (dapm_assert_locked.isra.36) from [<805842ec>] (dapm_mark_dirty+0x64/0xa4) [ 20.113165] [<80584288>] (dapm_mark_dirty) from [<805853a8>] (soc_dapm_dai_stream_event.isra.42+0x30/0xc8) [ 20.122863] r8:ed9b5dbc r7:00000000 r6:00000001 r5:00000001 r4:ed83d6c0 [ 20.129706] [<80585378>] (soc_dapm_dai_stream_event.isra.42) from [<80587e28>] (snd_soc_dapm_stream_event+0x78/0xa0) [ 20.140264] r5:ee2ee62c r4:00000001 [ 20.143918] [<80587db0>] (snd_soc_dapm_stream_event) from [<8058957c>] (soc_pcm_prepare+0x138/0x21c) [ 20.153058] r8:ed8d9480 r7:00000000 r6:ed9b0e00 r5:00000001 r4:ee2ee62c r3:00000000 ... This patch is to not probe the dummy component in soc_probe_component. Then there is no widget created for dummy DAI, and also don't need to check the dummy dai in dapm_connect_dai_link_widgets(). Signed-off-by: Shengjiu Wang <shengjiu.wang(a)freescale.com> --- changes in v2 - don't probe the component which is dummy - remove the dummy check in dapm_connect_dai_link_widgets(). sound/soc/soc-core.c | 2 +- sound/soc/soc-dapm.c | 5 ----- 2 files changed, 1 insertion(+), 6 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3a4a5c0..6ce6217 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1102,7 +1102,7 @@ static int soc_probe_component(struct snd_soc_card *card, struct snd_soc_dai *dai; int ret; - if (component->probed) + if (!strcmp(component->name, "snd-soc-dummy") || component->probed) return 0; component->card = card; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index aa327c9..37ab6b9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3820,11 +3820,6 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, for (i = 0; i < rtd->num_codecs; i++) { struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; - /* there is no point in connecting BE DAI links with dummies */ - if (snd_soc_dai_is_dummy(codec_dai) || - snd_soc_dai_is_dummy(cpu_dai)) - continue; - /* connect BE DAI playback if widgets are valid */ if (codec_dai->playback_widget && cpu_dai->playback_widget) { source = cpu_dai->playback_widget; -- 1.7.9.5

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[alsa-devel] Rane SL-4 USB sound card
by Tuukka Pasanen 07 Jul '15

07 Jul '15

Hello, I've been hacking to get Rane SL-4 USB audio (http://dj.rane.com/products/sl4-for-serato-scratch-live) which have 4 Input and 4 output channel to work with ALSA. After digging the lsusb and snoops I have separated endpoints for input (0x82) and output (0x06). They are just USB Isochronous I/O. I have compiled kernel and tried to get USB-quirks to work with this but then I have to admit that even Reloop Play seems be very similar it doesn't do the trick (No error but card doesn't come up eather) nor using Ranel SL-1 approach which just shoots the error [ 3555.009349] usb 1-5: Audio class v2 interfaces need an interface association [ 3555.009373] snd-usb-audio: probe of 1-5:1.0 failed with error -5 [ 3555.009394] usb 1-5: cannot find UAC_HEADER [ 3555.009401] snd-usb-audio: probe of 1-5:1.1 failed with error -5 [ 3555.009421] usb 1-5: cannot find UAC_HEADER [ 3555.009427] snd-usb-audio: probe of 1-5:1.2 failed with error -5 If I need interface association how to do that? Trying to adjust mixer values in mixer_maps.c. If someone have a time to give some guidance if this doable with Quirks or do I have to write some more specific driver? I attach lsusb below here. HID device seems to be adjusting buffer size and 48 Khz and 96 Khz output. Bus 001 Device 003: ID 1cc5:000f Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 255 bDeviceProtocol 255 bMaxPacketSize0 64 idVendor 0x1cc5 idProduct 0x000f bcdDevice 1.00 iManufacturer 1 Rane Corporation iProduct 2 Rane SL 4 iSerial 3 RH.01.00 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 235 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 255 Vendor Specific Class bInterfaceSubClass 1 bInterfaceProtocol 32 iInterface 4 SL 4 Audio ** UNRECOGNIZED: 09 24 01 00 02 0a 4b 00 00 ** UNRECOGNIZED: 08 24 0a 05 01 01 00 00 ** UNRECOGNIZED: 11 24 02 01 01 01 00 05 0a 00 00 00 00 05 00 00 00 ** UNRECOGNIZED: 0c 24 03 02 02 06 00 01 05 00 00 00 ** UNRECOGNIZED: 11 24 02 03 02 06 00 05 0a 00 00 00 00 0f 00 00 00 ** UNRECOGNIZED: 0c 24 03 04 01 01 00 03 05 00 00 00 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 32 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 32 iInterface 0 ** UNRECOGNIZED: 10 24 01 01 00 01 01 00 00 00 0a 00 00 00 00 05 ** UNRECOGNIZED: 06 24 02 01 04 18 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x06 EP 6 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0208 1x 520 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 32 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 32 iInterface 0 ** UNRECOGNIZED: 10 24 01 03 00 01 01 00 00 00 0a 00 00 00 00 0f ** UNRECOGNIZED: 06 24 02 01 04 18 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 37 Transfer Type Isochronous Synch Type Asynchronous Usage Type Implicit feedback Data wMaxPacketSize 0x0208 1x 520 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 3 Human Interface Device bInterfaceSubClass 0 No Subclass bInterfaceProtocol 0 None iInterface 0 HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 1.10 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 28 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0040 1x 64 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0040 1x 64 bytes bInterval 1 Device Qualifier (for other device speed): bLength 10 bDescriptorType 6 bcdUSB 2.00 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 255 bDeviceProtocol 255 bMaxPacketSize0 64 bNumConfigurations 1 Device Status: 0x0000 (Bus Powered) Thanks you for your time and any tips would be nice. Sincerely, Tuukka

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[alsa-devel] [PATCH] ASoC: topology: Fix TLV size calculation.
by Liam Girdwood 07 Jul '15

07 Jul '15

TLV size calculation was incorrectly calculated. Fix this according to include/sound/tlv.h Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/soc-topology.c | 21 ++++++++++----------- 1 file changed, 10 insertions(+), 11 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index d096068..7a19df3 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -580,27 +580,26 @@ static int soc_tplg_init_kcontrol(struct soc_tplg *tplg, } static int soc_tplg_create_tlv(struct soc_tplg *tplg, - struct snd_kcontrol_new *kc, u32 tlv_size) + struct snd_kcontrol_new *kc, struct snd_soc_tplg_ctl_tlv *tplg_tlv) { - struct snd_soc_tplg_ctl_tlv *tplg_tlv; struct snd_ctl_tlv *tlv; + int size; - if (tlv_size == 0) + if (tplg_tlv->count == 0) return 0; - tplg_tlv = (struct snd_soc_tplg_ctl_tlv *) tplg->pos; - tplg->pos += tlv_size; - - tlv = kzalloc(sizeof(*tlv) + tlv_size, GFP_KERNEL); + size = ((tplg_tlv->count + (sizeof(unsigned int) - 1)) & + ~(sizeof(unsigned int) - 1)); + tlv = kzalloc(sizeof(*tlv) + size, GFP_KERNEL); if (tlv == NULL) return -ENOMEM; dev_dbg(tplg->dev, " created TLV type %d size %d bytes\n", - tplg_tlv->numid, tplg_tlv->size); + tplg_tlv->numid, size); tlv->numid = tplg_tlv->numid; - tlv->length = tplg_tlv->size; - memcpy(tlv->tlv, tplg_tlv + 1, tplg_tlv->size); + tlv->length = size; + memcpy(&tlv->tlv[0], tplg_tlv->data, size); kc->tlv.p = (void *)tlv; return 0; @@ -773,7 +772,7 @@ static int soc_tplg_dmixer_create(struct soc_tplg *tplg, unsigned int count, } /* create any TLV data */ - soc_tplg_create_tlv(tplg, &kc, mc->hdr.tlv_size); + soc_tplg_create_tlv(tplg, &kc, &mc->tlv); /* register control here */ err = soc_tplg_add_kcontrol(tplg, &kc, -- 2.1.4

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[alsa-devel] [PATCH 0/2] ASoc: simple card: extend for MCLK support
by Arnaud Pouliquen 07 Jul '15

07 Jul '15

Patches propose to extend the management of the mclk-fs ratio. This extension allows to support following card constraints: - mclk is generated by cpu_dai, depending on frame rate. cpu_dai needs to be informed on the ratio supported by codec. - Each codec needs a specific ratio. In this case mclk-fs needs to be defined for the dai_link. Arnaud Pouliquen (2): ASoC: simple card: Add mclk-fs property in dai-link ASoC: simple card: set cpu-dai sysclk with mclk-fs .../devicetree/bindings/sound/simple-card.txt | 6 ++++- sound/soc/generic/simple-card.c | 27 +++++++++++++++++++--- 2 files changed, 29 insertions(+), 4 deletions(-) -- 1.9.1

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