October 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v2 5/6] ASoC: sirf-inner: add mach driver for SiRFSoC internal codec
by Barry Song 04 Nov '13

04 Nov '13

From: Rongjun Ying <Rongjun.Ying(a)csr.com> This connects DMA, CPU DAI and Codec DAI together and works as a mach driver. Signed-off-by: Rongjun Ying <Rongjun.Ying(a)csr.com> Signed-off-by: Barry Song <Baohua.Song(a)csr.com> --- -v2: use extcon for both Linux and Android support; use DAPM to replace legacy kcontrol sound/soc/sirf/Kconfig | 5 + sound/soc/sirf/Makefile | 2 + sound/soc/sirf/sirf-inner.c | 237 ++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 244 insertions(+) create mode 100644 sound/soc/sirf/sirf-inner.c diff --git a/sound/soc/sirf/Kconfig b/sound/soc/sirf/Kconfig index afa3952..564b0ec 100644 --- a/sound/soc/sirf/Kconfig +++ b/sound/soc/sirf/Kconfig @@ -11,3 +11,8 @@ config SND_SIRF_SOC_INNER config SND_SOC_SIRF_USP tristate + +config SND_SIRF_INNER + tristate "SoC Audio support for SiRF inner codec of SiRF EVB" + depends on SND_SIRF_SOC + select SND_SIRF_SOC_INNER diff --git a/sound/soc/sirf/Makefile b/sound/soc/sirf/Makefile index 8517c67..80abdf6 100644 --- a/sound/soc/sirf/Makefile +++ b/sound/soc/sirf/Makefile @@ -1,9 +1,11 @@ snd-soc-sirf-objs := sirf-pcm.o +snd-soc-sirf-inner-objs := sirf-inner.o snd-soc-sirf-soc-inner-objs := sirf-soc-inner.o snd-soc-sirf-i2s-objs := sirf-i2s.o snd-soc-sirf-usp-objs := sirf-usp.o obj-$(CONFIG_SND_SIRF_SOC) += snd-soc-sirf.o +obj-$(CONFIG_SND_SIRF_INNER) += snd-soc-sirf-inner.o obj-$(CONFIG_SND_SIRF_SOC_INNER) += snd-soc-sirf-soc-inner.o obj-$(CONFIG_SND_SOC_SIRF_I2S) += snd-soc-sirf-i2s.o obj-$(CONFIG_SND_SOC_SIRF_USP) += snd-soc-sirf-usp.o diff --git a/sound/soc/sirf/sirf-inner.c b/sound/soc/sirf/sirf-inner.c new file mode 100644 index 0000000..e36ac5e --- /dev/null +++ b/sound/soc/sirf/sirf-inner.c @@ -0,0 +1,237 @@ +/* + * SiRF inner audio device driver + * + * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company. + * + * Licensed under GPLv2 or later. + */ + +#include <linux/platform_device.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> + +#include <linux/extcon.h> +#include <linux/extcon/extcon-gpio.h> + +#define SIRF_JACK_GPIO_DEBOUNCE_TIME 200 /* in ms */ +struct sirf_inner_extcon_info { + struct platform_device pdev; + struct gpio_extcon_platform_data extcon_data; + int last_state; + int state_changed; +}; + +struct sirf_inner_card { + unsigned int gpio_hp_pa; + unsigned int gpio_spk_pa; + struct platform_device *sirf_inner_device; + struct sirf_inner_extcon_info extcon_info; +}; + +static int sirf_inner_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *ctrl, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct sirf_inner_card *sinner_card = snd_soc_card_get_drvdata(card); + int on = !SND_SOC_DAPM_EVENT_OFF(event); + if (gpio_is_valid(sinner_card->gpio_hp_pa)) + gpio_direction_output(sinner_card->gpio_hp_pa, on); + return 0; +} + +static int sirf_inner_spk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *ctrl, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct sirf_inner_card *sinner_card = snd_soc_card_get_drvdata(card); + int on = !SND_SOC_DAPM_EVENT_OFF(event); + + if (sinner_card->extcon_info.state_changed) { + sinner_card->extcon_info.state_changed = 0; + if (!sinner_card->extcon_info.last_state) + gpio_direction_output(sinner_card->gpio_spk_pa, 0); + else + gpio_direction_output(sinner_card->gpio_spk_pa, 1); + } else { + if (gpio_is_valid(sinner_card->gpio_spk_pa)) + gpio_direction_output(sinner_card->gpio_spk_pa, on); + } + + return 0; +} +static const struct snd_soc_dapm_widget sirf_inner_dapm_widgets[] = { + SND_SOC_DAPM_HP("Hp", sirf_inner_hp_event), + SND_SOC_DAPM_SPK("Ext Spk", sirf_inner_spk_event), + SND_SOC_DAPM_MIC("Ext Mic", NULL), +}; + +static const struct snd_soc_dapm_route intercon[] = { + {"Hp", NULL, "HPOUTL"}, + {"Hp", NULL, "HPOUTR"}, + {"Ext Spk", NULL, "SPKOUT"}, + {"MICIN1", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Ext Mic"}, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link sirf_inner_dai_links[] = { + { + .name = "SiRF inner", + .stream_name = "SiRF inner", + .codec_dai_name = "sirf-soc-inner", + .platform_name = "sirf-pcm-audio.1", + }, +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_sirf_inner_card = { + .name = "SiRF inner", + .owner = THIS_MODULE, + .dai_link = sirf_inner_dai_links, + .num_links = ARRAY_SIZE(sirf_inner_dai_links), + .dapm_widgets = sirf_inner_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(sirf_inner_dapm_widgets), + .dapm_routes = intercon, + .num_dapm_routes = ARRAY_SIZE(intercon), +}; + +static int sirf_inner_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &snd_soc_sirf_inner_card; + struct sirf_inner_card *sinner_card; + int ret; + + sinner_card = devm_kzalloc(&pdev->dev, sizeof(struct sirf_inner_card), + GFP_KERNEL); + if (sinner_card == NULL) + return -ENOMEM; + + sinner_card->sirf_inner_device = platform_device_alloc("soc-audio", -1); + if (!sinner_card->sirf_inner_device) + return -ENOMEM; + + sirf_inner_dai_links[0].cpu_of_node = + of_parse_phandle(pdev->dev.of_node, "sirf,inner-platform", 0); + sirf_inner_dai_links[0].codec_of_node = + of_parse_phandle(pdev->dev.of_node, "sirf,inner-codec", 0); + sinner_card->gpio_spk_pa = of_get_named_gpio(pdev->dev.of_node, + "spk-pa-gpios", 0); + sinner_card->gpio_hp_pa = of_get_named_gpio(pdev->dev.of_node, + "hp-pa-gpios", 0); + if (gpio_is_valid(sinner_card->gpio_spk_pa)) + gpio_request(sinner_card->gpio_spk_pa, "SPA_PA_SD"); + if (gpio_is_valid(sinner_card->gpio_hp_pa)) + gpio_request(sinner_card->gpio_hp_pa, "HP_PA_SD"); + + card->dev = &pdev->dev; + snd_soc_card_set_drvdata(card, sinner_card); + platform_set_drvdata(pdev, card); + if (gpio_is_valid(sinner_card->gpio_hp_pa)) + gpio_direction_output(sinner_card->gpio_hp_pa, 0); + if (gpio_is_valid(sinner_card->gpio_spk_pa)) + gpio_direction_output(sinner_card->gpio_spk_pa, 0); + + platform_set_drvdata(sinner_card->sirf_inner_device, + &snd_soc_sirf_inner_card); + + ret = platform_device_add(sinner_card->sirf_inner_device); + if (ret) { + platform_device_put(sinner_card->sirf_inner_device); + return ret; + } + + sinner_card->extcon_info.extcon_data.name = "h2w"; + sinner_card->extcon_info.extcon_data.debounce = + SIRF_JACK_GPIO_DEBOUNCE_TIME; + sinner_card->extcon_info.extcon_data.irq_flags = + IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_SHARED; + sinner_card->extcon_info.extcon_data.state_on = "0"; + sinner_card->extcon_info.extcon_data.state_off = "1"; + sinner_card->extcon_info.extcon_data.gpio = + of_get_named_gpio(pdev->dev.of_node, + "hp-switch-gpios", 0); + + sinner_card->extcon_info.pdev.name = "extcon-gpio"; + sinner_card->extcon_info.pdev.id = pdev->id; + sinner_card->extcon_info.pdev.dev.platform_data = + &sinner_card->extcon_info.extcon_data; + + return platform_device_register(&sinner_card->extcon_info.pdev); +} + +static int sirf_inner_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct sirf_inner_card *sinner_card = snd_soc_card_get_drvdata(card); + + if (gpio_is_valid(sinner_card->gpio_hp_pa)) + gpio_free(sinner_card->gpio_hp_pa); + if (gpio_is_valid(sinner_card->gpio_spk_pa)) + gpio_free(sinner_card->gpio_spk_pa); + + platform_device_put(sinner_card->sirf_inner_device); + return 0; +} + +#ifdef CONFIG_PM_SLEEP +static int sirf_inner_resume(struct device *dev) +{ + struct snd_soc_card *card = dev_get_drvdata(dev); + struct sirf_inner_card *sinner_card = snd_soc_card_get_drvdata(card); + struct extcon_dev *edev; + int state; + + edev = extcon_get_extcon_dev(sinner_card->extcon_info.extcon_data.name); + state = gpio_get_value(sinner_card->extcon_info.extcon_data.gpio); + if (state != sinner_card->extcon_info.last_state) { + sinner_card->extcon_info.state_changed = 1; + sinner_card->extcon_info.last_state = state; + extcon_set_state(edev, state); + } + return 0; +} + +static int sirf_inner_suspend(struct device *dev) +{ + struct snd_soc_card *card = dev_get_drvdata(dev); + struct sirf_inner_card *sinner_card = snd_soc_card_get_drvdata(card); + sinner_card->extcon_info.last_state = gpio_get_value(sinner_card->extcon_info.extcon_data.gpio); + if (gpio_is_valid(sinner_card->gpio_spk_pa)) + gpio_direction_output(sinner_card->gpio_spk_pa, 0); + return 0; +} +#endif + +static const struct of_device_id sirf_inner_of_match[] = { + {.compatible = "sirf,sirf-inner", }, + { }, +}; +MODULE_DEVICE_TABLE(of, sirf_inner_of_match); + +static const struct dev_pm_ops sirf_inner_pm_ops = { + SET_SYSTEM_SLEEP_PM_OPS(sirf_inner_suspend, sirf_inner_resume) +}; + +static struct platform_driver sirf_inner_driver = { + .driver = { + .name = "sirf-inner", + .owner = THIS_MODULE, + .pm = &sirf_inner_pm_ops, + .of_match_table = sirf_inner_of_match, + }, + .probe = sirf_inner_probe, + .remove = sirf_inner_remove, +}; +module_platform_driver(sirf_inner_driver); + +MODULE_AUTHOR("RongJun Ying <RongJun.Ying(a)csr.com>"); +MODULE_DESCRIPTION("ALSA SoC SIRF inner AUDIO driver"); +MODULE_LICENSE("GPL v2"); -- 1.8.2.3

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[alsa-devel] include/uapi/firewire.h for other firewire drivers
by Takashi Sakamoto 03 Nov '13

03 Nov '13

Hi Clemens, About include/uapi/firewire.h, I'd like you to give me an advice. I'm sorry but I have little knowledgement about Dice chipset. So here I want to know what I should do for snd-fireworks/snd-bebob. The header file includes an interface for users to know its already streaming or not, and lock it. But for Fireworks/BeBoB, I think checking CMP connections do the same thing. Current FFADO can be failed when CMP connections on the device are established, and ALSA can do the same. (but there are some problems. both drivers set sampling rate before checking CMP connections. For this I have a plan to post a patch for both FFADO and ALSA.) Should I implement the same lock for Fireworks/BeBoB drivers? Regards Takashi Sakamoto

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[alsa-devel] [PATCH] ASoC: dpcm: improve robustness
by Russell King - ARM Linux 01 Nov '13

01 Nov '13

Mark, Liam, You may also wish to consider adding this patch as well - it avoids oopsing due to a NULL be_substream when writing be_substream->runtime if the frontend is marked as having a playback/capture stream but there is no backend. I found this with Liam's "Explicitly set capture/playback support in DAI links" patch applied, and I'd configured the FE to have both capture and playback, but I'd forgotten to configure the BE appropriately. It seems nicer to gracefully handle this than oops the kernel. 8<===== From: Russell King <rmk+kernel(a)arm.linux.org.uk> ASoC: dpcm: improve robustness Avoid oopsing if there is no backend stream associated with a front end stream. Signed-off-by: Russell King <rmk+kernel(a)arm.linux.org.uk> --- sound/soc/soc-pcm.c | 9 ++++++++- 1 files changed, 8 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 7798072c45ea..419af79fa30e 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1036,6 +1036,12 @@ static int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream) struct snd_pcm_substream *be_substream = snd_soc_dpcm_get_substream(be, stream); + if (!be_substream) { + dev_err(be->dev, "ASoC: no backend %s stream\n", + stream ? "capture" : "playback"); + continue; + } + /* is this op for this BE ? */ if (!snd_soc_dpcm_be_can_update(fe, be, stream)) continue; @@ -1053,7 +1059,8 @@ static int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream) (be->dpcm[stream].state != SND_SOC_DPCM_STATE_CLOSE)) continue; - dev_dbg(be->dev, "ASoC: open BE %s\n", be->dai_link->name); + dev_dbg(be->dev, "ASoC: open %s BE %s\n", + stream ? "capture" : "playback", be->dai_link->name); be_substream->runtime = be->dpcm[stream].runtime; err = soc_pcm_open(be_substream); -- 1.7.4.4

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[alsa-devel] Audinst HUD-mx2 covertly muting itself
by Tobias Klausmann 01 Nov '13

01 Nov '13

Hey, I have an Audinst HUD-mx2 external USB audio device: Bus 007 Device 006: ID 040d:340f VIA Technologies, Inc. $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: HUDmx2 [Audinst HUD-mx2], device 0: USB Audio [USB Audio] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: HUDmx2 [Audinst HUD-mx2], device 1: USB Audio [USB Audio #1] Subdevices: 1/1 Subdevice #0: subdevice #0 While the device works right in principle, every time an app closes the device, it mutes itself (or crash, this may be a Hw problem). There is no indication of the broken state in the mixer displays of alsamixer or alsamixer-gui, and amixer output does not indicate any change, either (see below). I am a bit stumped as to what to do next. The vendor does not officially support anything but Windows, so getting help from them is very unlikely. amixer output in working/not working/working again condition: ~ $ amixer -c0 contents # working numid=1,iface=MIXER,name='PCM Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=2,iface=MIXER,name='PCM Playback Switch',index=1 ; type=BOOLEAN,access=rw------,values=1 : values=on numid=3,iface=MIXER,name='PCM Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=45,step=0 : values=45,45 | dBminmax-min=42949627.96dB,max=0.00dB ~ $ amixer -c0 contents # not working numid=1,iface=MIXER,name='PCM Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=2,iface=MIXER,name='PCM Playback Switch',index=1 ; type=BOOLEAN,access=rw------,values=1 : values=on numid=3,iface=MIXER,name='PCM Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=45,step=0 : values=45,45 | dBminmax-min=42949627.96dB,max=0.00dB ~ $ amixer -c0 contents # working again numid=1,iface=MIXER,name='PCM Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=2,iface=MIXER,name='PCM Playback Switch',index=1 ; type=BOOLEAN,access=rw------,values=1 : values=on numid=3,iface=MIXER,name='PCM Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=45,step=0 : values=45,45 | dBminmax-min=42949627.96dB,max=0.00dB Kernel version is 3.2.5 (Ubuntu). $ cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.24. $ cat /proc/asound/cards 0 [HUDmx2 ]: USB-Audio - Audinst HUD-mx2 VIA Technologies Inc. Audinst HUD-mx2 at usb-0000:00:1d.1-2, full speed $ cat /proc/asound/modules 0 snd_usb_audio $ cat /proc/asound/devices 1: : sequencer 2: [ 0- 1]: digital audio playback 3: [ 0- 0]: digital audio playback 4: [ 0] : control 33: : timer $ lsusb -v [...] Bus 007 Device 006: ID 040d:340f VIA Technologies, Inc. Couldn't open device, some information will be missing Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 64 idVendor 0x040d VIA Technologies, Inc. idProduct 0x340f bcdDevice 1.01 iManufacturer 1 iProduct 2 iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 255 bNumInterfaces 3 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xe0 Self Powered Remote Wakeup MaxPower 0mA 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 9 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 59 bInCollection 1 baInterfaceNr( 0) 1 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 12 wTerminalType 0x0605 SPDIF interface bAssocTerminal 0 bSourceID 11 iTerminal 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 11 bSourceID 1 bControlSize 1 bmaControls( 0) 0x01 Mute Control bmaControls( 1) 0x00 bmaControls( 2) 0x00 iFeature 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0301 Speaker bAssocTerminal 0 bSourceID 5 iTerminal 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 5 bSourceID 1 bControlSize 1 bmaControls( 0) 0x01 Mute Control bmaControls( 1) 0x02 Volume Control bmaControls( 2) 0x02 Volume Control iFeature 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 1 bDelay 1 frames wFormatTag 8193 IEC1937_AC-3 AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 3 (FORMAT_TYPE_III) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 2 Discrete tSamFreq[ 0] 44100 tSamFreq[ 1] 48000 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x00c0 1x 192 bytes bInterval 1 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 1 Milliseconds wLockDelay 1 Milliseconds Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 20 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 4 Discrete tSamFreq[ 0] 44100 tSamFreq[ 1] 48000 tSamFreq[ 2] 88200 tSamFreq[ 3] 96000 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0180 1x 384 bytes bInterval 1 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 1 Milliseconds wLockDelay 1 Milliseconds Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 3 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 20 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 4 Discrete tSamFreq[ 0] 44100 tSamFreq[ 1] 48000 tSamFreq[ 2] 88200 tSamFreq[ 3] 96000 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0240 1x 576 bytes bInterval 1 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 1 Milliseconds wLockDelay 1 Milliseconds Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 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 61 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x8f EP 15 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0010 1x 16 bytes bInterval 16 [...] Any help is appreciated. Regards, Tobias -- Aussen Stop-Bits, innen gepackt!

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[alsa-devel] [PATCH 1/3] ASoC: wm_adsp: Print error when regmap reads/writes fail
by Dimitris Papastamos 01 Nov '13

01 Nov '13

Change-Id: I782f7684d9ef38adcbee6c6ed6809534dc7ff9b9 Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index e09ac94..cd007cc 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -339,8 +339,8 @@ static int wm_coeff_write_control(struct snd_kcontrol *kcontrol, ret = regmap_raw_write(adsp->regmap, reg, scratch, ctl->len); if (ret) { - adsp_err(adsp, "Failed to write %zu bytes to %x\n", - ctl->len, reg); + adsp_err(adsp, "Failed to write %zu bytes to %x: %d\n", + ctl->len, reg, ret); kfree(scratch); return ret; } @@ -393,8 +393,8 @@ static int wm_coeff_read_control(struct snd_kcontrol *kcontrol, ret = regmap_raw_read(adsp->regmap, reg, scratch, ctl->len); if (ret) { - adsp_err(adsp, "Failed to read %zu bytes from %x\n", - ctl->len, reg); + adsp_err(adsp, "Failed to read %zu bytes from %x: %d\n", + ctl->len, reg, ret); kfree(scratch); return ret; } @@ -1227,8 +1227,8 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) le32_to_cpu(blk->len)); if (ret != 0) { adsp_err(dsp, - "%s.%d: Failed to write to %x in %s\n", - file, blocks, reg, region_name); + "%s.%d: Failed to write to %x in %s: %d\n", + file, blocks, reg, region_name, ret); } kfree(buf); -- 1.8.4.2

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[alsa-devel] [PATCH] ALSA: hda - not choose assigned converters for unused pins of Valleyview
by mengdong.lin＠intel.com 01 Nov '13

01 Nov '13

From: Mengdong Lin <mengdong.lin(a)intel.com> For Valleyview display codec, if an unused pin chooses an assgined converter selected by a used pin, playback on the unused pin can also give sound to the output device of the used pin. It's because data flows from the same convertor to the display port of the used pin. This issue is same as Haswell. So this patch avoids using assinged convertors for unused pins. The related function haswell_config_cvts() is renamed for code reuse. 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 e332db8..bcf764f 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -45,6 +45,7 @@ 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_valleyview(codec) ((codec)->vendor_id == 0x80862882) struct hdmi_spec_per_cvt { hda_nid_t cvt_nid; @@ -1148,7 +1149,7 @@ static int hdmi_choose_cvt(struct hda_codec *codec, return 0; } -static void haswell_config_cvts(struct hda_codec *codec, +static void not_share_unassigned_cvt(struct hda_codec *codec, hda_nid_t pin_nid, int mux_idx) { struct hdmi_spec *spec = codec->spec; @@ -1226,8 +1227,8 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, mux_idx); /* configure unused pins to choose other converters */ - if (is_haswell(codec)) - haswell_config_cvts(codec, per_pin->pin_nid, mux_idx); + if (is_haswell(codec) || is_valleyview(codec)) + not_share_unassigned_cvt(codec, per_pin->pin_nid, mux_idx); snd_hda_spdif_ctls_assign(codec, pin_idx, per_cvt->cvt_nid); -- 1.8.1.2

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[alsa-devel] [PATCH v4 0/4] Add dual-fifo mode support of i.MX ssi
by Nicolin Chen 01 Nov '13

01 Nov '13

Changelog v4: * PATCH-3: Drop useless register configuration. * * Nothing changes for the other three patches v3: * PATCH-1: Add comments to indicate the end of v1 and v2 array. * PATCH-3: Use better way to keep watermark as even number. * * Nothing changes for PATCH-2 and PATCH-4 v2: * Instead of adding rogue scripts to current SDMA driver based on firmware * V1, we define the new SDMA firmware as version 2 and bisect the PATCH-1 * to two patches: The first is to add version check code to the SDMA driver; * And the second is to add SSI dual FIFO DMATYPE. * * Nothing changes for the last two patches. v1: * SSI can reduce hardware overrun/underrun possibility when using dual * fifo mode. To support this mode, we need to first update sdma sciprt * list, and then enable dual fifo BIT in SSI driver, and last update DT * bindings of i.MX series. * * ! This series of patches has a direct dependency between them. When * ! applying them, we need to apply to one single branch. Otherwise, * ! it would break currect branches. Nicolin Chen (4): dma: imx-sdma: Add sdma firmware version 2 support dma: imx-sdma: Add new dma type for ssi dual fifo script ASoC: fsl_ssi: Add dual fifo mode support ARM: dts: imx: use dual-fifo sdma script for ssi .../devicetree/bindings/dma/fsl-imx-sdma.txt | 1 + arch/arm/boot/dts/imx51.dtsi | 4 ++-- arch/arm/boot/dts/imx53.dtsi | 4 ++-- arch/arm/boot/dts/imx6qdl.dtsi | 12 ++++++------ arch/arm/boot/dts/imx6sl.dtsi | 12 ++++++------ drivers/dma/imx-sdma.c | 19 ++++++++++++++++++- include/linux/platform_data/dma-imx-sdma.h | 5 +++++ include/linux/platform_data/dma-imx.h | 1 + sound/soc/fsl/fsl_ssi.c | 18 +++++++++++++++++- 9 files changed, 58 insertions(+), 18 deletions(-) -- 1.8.4

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[alsa-devel] [PATCHv7][ 1/5] ASoC: eukrea-tlv320: Use dev_err instead of pr_err.
by Denis Carikli 31 Oct '13

31 Oct '13

It also contains a minor style cleanup. Cc: Rob Herring <rob.herring(a)calxeda.com> Cc: Pawel Moll <pawel.moll(a)arm.com> Cc: Mark Rutland <mark.rutland(a)arm.com> Cc: Stephen Warren <swarren(a)wwwdotorg.org> Cc: Ian Campbell <ijc+devicetree(a)hellion.org.uk> Cc: devicetree(a)vger.kernel.org Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Cc: Sascha Hauer <kernel(a)pengutronix.de> Cc: linux-arm-kernel(a)lists.infradead.org Cc: Eric Bénard <eric(a)eukrea.com> Signed-off-by: Denis Carikli <denis(a)eukrea.com> --- ChangeLog v6->v7: - eukrea-tlv320 cleanups are now moved into that patch. --- sound/soc/fsl/eukrea-tlv320.c | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index 9a4a0ca..5983740 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -42,7 +42,8 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret) { - pr_err("%s: failed set cpu dai format\n", __func__); + dev_err(cpu_dai->dev, + "Failed to set the cpu dai format.\n"); return ret; } @@ -50,14 +51,16 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret) { - pr_err("%s: failed set codec dai format\n", __func__); + dev_err(cpu_dai->dev, + "Failed to set the codec format.\n"); return ret; } ret = snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_OUT); if (ret) { - pr_err("%s: failed setting codec sysclk\n", __func__); + dev_err(cpu_dai->dev, + "Failed to set the codec sysclk.\n"); return ret; } snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0); @@ -65,7 +68,8 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0, SND_SOC_CLOCK_IN); if (ret) { - pr_err("can't set CPU system clock IMX_SSP_SYS_CLK\n"); + dev_err(cpu_dai->dev, + "Can't set the IMX_SSP_SYS_CLK CPU system clock.\n"); return ret; } @@ -155,7 +159,8 @@ static struct platform_driver eukrea_tlv320_driver = { .owner = THIS_MODULE, }, .probe = eukrea_tlv320_probe, - .remove = eukrea_tlv320_remove,}; + .remove = eukrea_tlv320_remove, +}; module_platform_driver(eukrea_tlv320_driver); -- 1.7.9.5

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[alsa-devel] [RFC PATCH] ASoC: sgtl5000: interconnect ADC with Mic Bias
by Hector Palacios 31 Oct '13

31 Oct '13

The interconnection of the Mic Bias to the ADC allows to control the Mic Bias Output Impedance. Without it, recording from a PC microphone results in silence. Signed-off-by: Hector Palacios <hector.palacios(a)digi.com> --- sound/soc/codecs/sgtl5000.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index ea47938..60d5012 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -231,6 +231,7 @@ static const struct snd_soc_dapm_route sgtl5000_dapm_routes[] = { {"Capture Mux", "MIC_IN", "MIC_IN"}, /* mic_in --> adc_mux */ {"ADC", NULL, "VAG_POWER"}, + {"ADC", NULL, "Mic Bias"}, {"ADC", NULL, "Capture Mux"}, /* adc_mux --> adc */ {"AIFOUT", NULL, "ADC"}, /* adc --> i2s_out */

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[alsa-devel] [PATCH] ALSA: fix oops in snd_pcm_info() caused by ASoC DPCM
by Russell King - ARM Linux 31 Oct '13

31 Oct '13

Liam, Mark, As discussed privately, here's the fix for the ASoC oops I've been seeing with DPCM in ALSA. This avoids unnecessarily exposing the ASoC internal PCM devices to userspace, which is complimentary to your patch fixing the deref of the PCM ops. 8<======== From: Russell King <rmk+kernel(a)arm.linux.org.uk> Subject: [PATCH] ALSA: fix oops in snd_pcm_info() caused by ASoC DPCM Unable to handle kernel NULL pointer dereference at virtual address 00000008 pgd = d5300000 [00000008] *pgd=0d265831, *pte=00000000, *ppte=00000000 Internal error: Oops: 17 [#1] PREEMPT ARM CPU: 0 PID: 2295 Comm: vlc Not tainted 3.11.0+ #755 task: dee74800 ti: e213c000 task.ti: e213c000 PC is at snd_pcm_info+0xc8/0xd8 LR is at 0x30232065 pc : [<c031b52c>] lr : [<30232065>] psr: a0070013 sp : e213dea8 ip : d81cb0d0 fp : c05f7678 r10: c05f7770 r9 : fffffdfd r8 : 00000000 r7 : d8a968a8 r6 : d8a96800 r5 : d8a96200 r4 : d81cb000 r3 : 00000000 r2 : d81cb000 r1 : 00000001 r0 : d8a96200 Flags: NzCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment user Control: 10c5387d Table: 15300019 DAC: 00000015 Process vlc (pid: 2295, stack limit = 0xe213c248) [<c031b52c>] (snd_pcm_info) from [<c031b570>] (snd_pcm_info_user+0x34/0x9c) [<c031b570>] (snd_pcm_info_user) from [<c03164a4>] (snd_pcm_control_ioctl+0x274/0x280) [<c03164a4>] (snd_pcm_control_ioctl) from [<c0311458>] (snd_ctl_ioctl+0xc0/0x55c) [<c0311458>] (snd_ctl_ioctl) from [<c00eca84>] (do_vfs_ioctl+0x80/0x31c) [<c00eca84>] (do_vfs_ioctl) from [<c00ecd5c>] (SyS_ioctl+0x3c/0x60) [<c00ecd5c>] (SyS_ioctl) from [<c000e500>] (ret_fast_syscall+0x0/0x48) Code: e1a00005 e59530dc e3a01001 e1a02004 (e5933008) ---[ end trace cb3d9bdb8dfefb3c ]--- This is provoked when the ASoC front end is open along with its backend, (which causes the backend to have a runtime assigned to it) and then the SNDRV_CTL_IOCTL_PCM_INFO is requested for the (visible) backend device. Resolve this by ensuring that ASoC internal backend devices are not visible to userspace, just as the commentry for snd_pcm_new_internal() says it should be. Signed-off-by: Russell King <rmk+kernel(a)arm.linux.org.uk> --- sound/core/pcm.c | 4 ++++ 1 files changed, 4 insertions(+), 0 deletions(-) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 17f45e8aa89c..e1e9e0c999fe 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -49,6 +49,8 @@ static struct snd_pcm *snd_pcm_get(struct snd_card *card, int device) struct snd_pcm *pcm; list_for_each_entry(pcm, &snd_pcm_devices, list) { + if (pcm->internal) + continue; if (pcm->card == card && pcm->device == device) return pcm; } @@ -60,6 +62,8 @@ static int snd_pcm_next(struct snd_card *card, int device) struct snd_pcm *pcm; list_for_each_entry(pcm, &snd_pcm_devices, list) { + if (pcm->internal) + continue; if (pcm->card == card && pcm->device > device) return pcm->device; else if (pcm->card->number > card->number) -- 1.7.4.4

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