June 2012 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] M-Audio FastTrack C400
by Grant Diffey 18 Aug '12

18 Aug '12

So this looks reasonably familiar, int0 is DFU int1 is mixer int2 is audio out int3 is audio in int4 is midi int5 is HID Quirk should look like the fast track ultra basically. maybe the mixer is less broken in this device? Should we start with just QUIRK_STANDARD_AUDIO_INTERFACE or copy the whole data section from the Ultra quirk? This has isosyncronous feedback for clock control needs current sound tree usb-audio tree Grant On Wed, Apr 18, 2012 at 8:45 AM, Chris Cavey <chris-alsa(a)rauros.net> wrote: > Will to help with development, may need pointers to get started and > willing to test things. I am mostly concerned with simple playback. > Recording, DSP, and Midi can all come later if at all. > > As requested the output of sudo lsusb -v -d 0763:2030" > > Bus 002 Device 003: ID 0763:2030 Midiman > Device Descriptor: > bLength 18 > bDescriptorType 1 > bcdUSB 2.00 > bDeviceClass 239 Miscellaneous Device > bDeviceSubClass 2 ? > bDeviceProtocol 1 Interface Association > bMaxPacketSize0 64 > idVendor 0x0763 Midiman > idProduct 0x2030 > bcdDevice 1.01 > iManufacturer 1 M-Audio > iProduct 2 Fast Track C400 > iSerial 0 > bNumConfigurations 1 > Configuration Descriptor: > bLength 9 > bDescriptorType 2 > wTotalLength 455 > bNumInterfaces 6 > bConfigurationValue 1 > iConfiguration 0 > bmAttributes 0x80 > (Bus Powered) > MaxPower 500mA > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 0 > bAlternateSetting 0 > bNumEndpoints 0 > bInterfaceClass 254 Application Specific Interface > bInterfaceSubClass 1 Device Firmware Update > bInterfaceProtocol 0 > iInterface 6 Fast Track C400 DFU > Device Firmware Upgrade Interface Descriptor: > bLength 9 > bDescriptorType 33 > bmAttributes 1 > Will Not Detach > Manifestation Intolerant > Upload Unsupported > Download Supported > wDetachTimeout 2000 milliseconds > wTransferSize 1024 bytes > bcdDFUVersion 1.10 > Interface Association: > bLength 8 > bDescriptorType 11 > bFirstInterface 1 > bInterfaceCount 4 > bFunctionClass 255 Vendor Specific Class > bFunctionSubClass 0 > bFunctionProtocol 32 > iFunction 2 Fast Track C400 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 1 > bAlternateSetting 0 > bNumEndpoints 0 > bInterfaceClass 255 Vendor Specific Class > bInterfaceSubClass 1 > bInterfaceProtocol 32 > iInterface 2 Fast Track C400 > ** UNRECOGNIZED: 09 24 01 00 02 0a a3 00 00 > ** UNRECOGNIZED: 09 24 0b 80 02 81 82 03 00 > ** UNRECOGNIZED: 08 24 0a 81 03 03 03 00 > ** UNRECOGNIZED: 08 24 0a 82 00 05 03 00 > ** UNRECOGNIZED: 11 24 02 00 01 01 00 80 06 00 00 00 00 00 00 00 00 > ** UNRECOGNIZED: 0c 24 03 01 01 01 00 02 80 00 00 00 > ** UNRECOGNIZED: 11 24 02 02 03 06 00 80 04 00 00 00 00 00 00 00 00 > ** UNRECOGNIZED: 0c 24 03 03 03 06 00 40 80 00 00 00 > ** UNRECOGNIZED: 11 24 09 20 00 00 02 02 00 04 00 00 00 00 00 00 00 > ** UNRECOGNIZED: 15 24 04 40 03 02 00 43 04 00 00 00 00 00 ff ff ff > ff 0f 00 00 > ** UNRECOGNIZED: 10 24 04 42 02 02 00 01 00 00 00 00 00 ff 00 00 > ** UNRECOGNIZED: 10 24 07 43 02 00 42 fd 03 00 00 fd 03 00 00 00 > 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 4 Line > ** UNRECOGNIZED: 10 24 01 00 00 01 01 00 00 00 06 00 00 00 00 00 > ** UNRECOGNIZED: 06 24 02 01 03 18 > Endpoint Descriptor: > bLength 7 > bDescriptorType 5 > bEndpointAddress 0x01 EP 1 OUT > bmAttributes 37 > Transfer Type Isochronous > Synch Type Asynchronous > Usage Type Implicit feedback Data > wMaxPacketSize 0x00ea 1x 234 bytes > bInterval 1 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 2 > bAlternateSetting 2 > bNumEndpoints 1 > bInterfaceClass 255 Vendor Specific Class > bInterfaceSubClass 2 > bInterfaceProtocol 32 > iInterface 4 Line > ** UNRECOGNIZED: 10 24 01 00 00 01 d0 00 00 00 02 00 00 00 00 00 > ** UNRECOGNIZED: 06 24 02 03 02 10 > Endpoint Descriptor: > bLength 7 > bDescriptorType 5 > bEndpointAddress 0x01 EP 1 OUT > bmAttributes 37 > Transfer Type Isochronous > Synch Type Asynchronous > Usage Type Implicit feedback Data > wMaxPacketSize 0x004e 1x 78 bytes > bInterval 1 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 3 > bAlternateSetting 0 > bNumEndpoints 0 > bInterfaceClass 255 Vendor Specific Class > bInterfaceSubClass 2 > bInterfaceProtocol 32 > iInterface 4 Line > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 3 > bAlternateSetting 1 > bNumEndpoints 1 > bInterfaceClass 255 Vendor Specific Class > bInterfaceSubClass 2 > bInterfaceProtocol 32 > iInterface 4 Line > ** UNRECOGNIZED: 10 24 01 00 00 01 01 00 00 00 04 00 00 00 00 00 > ** UNRECOGNIZED: 06 24 02 01 03 18 > Endpoint Descriptor: > bLength 7 > bDescriptorType 5 > bEndpointAddress 0x81 EP 1 IN > bmAttributes 5 > Transfer Type Isochronous > Synch Type Asynchronous > Usage Type Data > wMaxPacketSize 0x009c 1x 156 bytes > bInterval 1 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 4 > bAlternateSetting 0 > bNumEndpoints 2 > bInterfaceClass 1 Audio > bInterfaceSubClass 3 MIDI Streaming > bInterfaceProtocol 0 > iInterface 0 > MIDIStreaming Interface Descriptor: > bLength 7 > bDescriptorType 36 > bDescriptorSubtype 1 (HEADER) > bcdADC 1.00 > wTotalLength 35 > MIDIStreaming Interface Descriptor: > bLength 6 > bDescriptorType 36 > bDescriptorSubtype 2 (MIDI_IN_JACK) > bJackType 1 Embedded > bJackID 1 > iJack 0 > MIDIStreaming Interface Descriptor: > bLength 6 > bDescriptorType 36 > bDescriptorSubtype 2 (MIDI_IN_JACK) > bJackType 2 External > bJackID 2 > iJack 0 > MIDIStreaming Interface Descriptor: > bLength 8 > bDescriptorType 36 > bDescriptorSubtype 3 (MIDI_OUT_JACK) > Warning: Descriptor too short > bJackType 1 Embedded > bJackID 3 > bNrInputPins 1 > baSourceID( 0) 2 > BaSourcePin( 0) 1 > iJack 8 > MIDIStreaming Interface Descriptor: > bLength 8 > bDescriptorType 36 > bDescriptorSubtype 3 (MIDI_OUT_JACK) > Warning: Descriptor too short > bJackType 2 External > bJackID 4 > bNrInputPins 1 > baSourceID( 0) 1 > BaSourcePin( 0) 1 > iJack 105 > Endpoint Descriptor: > bLength 7 > bDescriptorType 5 > bEndpointAddress 0x07 EP 7 OUT > bmAttributes 2 > Transfer Type Bulk > Synch Type None > Usage Type Data > wMaxPacketSize 0x0200 1x 512 bytes > bInterval 0 > MIDIStreaming Endpoint Descriptor: > bLength 5 > bDescriptorType 37 > bDescriptorSubtype 1 (GENERAL) > bNumEmbMIDIJack 1 > baAssocJackID( 0) 1 > Endpoint Descriptor: > bLength 7 > bDescriptorType 5 > bEndpointAddress 0x87 EP 7 IN > bmAttributes 2 > Transfer Type Bulk > Synch Type None > Usage Type Data > wMaxPacketSize 0x0200 1x 512 bytes > bInterval 0 > MIDIStreaming Endpoint Descriptor: > bLength 5 > bDescriptorType 37 > bDescriptorSubtype 1 (GENERAL) > bNumEmbMIDIJack 1 > baAssocJackID( 0) 3 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 5 > bAlternateSetting 0 > bNumEndpoints 1 > bInterfaceClass 3 Human Interface Device > bInterfaceSubClass 0 No Subclass > bInterfaceProtocol 1 Keyboard > iInterface 0 > HID Device Descriptor: > bLength 9 > bDescriptorType 33 > bcdHID 1.10 > bCountryCode 0 Not supported > bNumDescriptors 1 > bDescriptorType 34 Report > wDescriptorLength 63 > Report Descriptors: > ** UNAVAILABLE ** > Endpoint Descriptor: > bLength 7 > bDescriptorType 5 > bEndpointAddress 0x86 EP 6 IN > bmAttributes 3 > Transfer Type Interrupt > Synch Type None > Usage Type Data > wMaxPacketSize 0x0008 1x 8 bytes > bInterval 8 > Device Qualifier (for other device speed): > bLength 10 > bDescriptorType 6 > bcdUSB 2.00 > bDeviceClass 239 Miscellaneous Device > bDeviceSubClass 2 ? > bDeviceProtocol 1 Interface Association > bMaxPacketSize0 64 > bNumConfigurations 1 > Device Status: 0x0000 > (Bus Powered) > > > On 04/17/2012 09:15 AM, Grant Diffey wrote: > > start with an lsusb -v -d (vid:pid) pasted here > > On Tue, Apr 17, 2012 at 10:37 PM, Chris Cavey <chris-alsa(a)rauros.net>wrote: > >> I'd love to be able to help ALSA support this USB sound device. I've >> currently got the device working in a VM on my Ubuntu 12.04 workstation. >> I believe I can capture USB packets. Would this be helpful to anyone? >> Any pointers on where to get started? >> _______________________________________________ >> Alsa-devel mailing list >> Alsa-devel(a)alsa-project.org >> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >> > > >

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[alsa-devel] Help requested: new HSS1394 MIDI back-end
by Sean M. Pappalardo - D.J. Pegasus 01 Aug '12

01 Aug '12

Hello everyone. I am a developer for Mixxx, GPL cross-platform digital DJ software: http://www.mixxx.org. My focus is the controller subsystem. I am ready to finally add Linux support for HSS1394 DJ controllers such as Stanton's SCS.1m and SCS.1d. HSS1394 is just MIDI over IEEE1394 (Firewire) at wire speed. The Windows and OSX library source code is LGPL and available on LaunchPad here: https://launchpad.net/hss1394 The Stanton SCS.1m is a sound interface integrated with a two-way control surface that speaks HSS1394. (The sound card uses an Oxford chip set and works fine with FFADO 2.0.1 today up to 96kHz. I use it with Mixxx via JACK.) The Stanton SCS.1d is a two-way control surface (no sound card) that speaks HSS1394 and is capable of sending ALOT of latency-sensitive data: about 4000 messages per platter rotation. (Now imagine if you had four or even two of them on the same network running at +50% speed. This would quickly overwhelm USB, at least at the time these devices were designed.) Read about them here: http://www.stantondj.com/stanton-controllers-systems/system1.html After a lengthy discussion on the FFADO mailing list, we determined that the best (cleanest and most flexible) way to get Linux support for these devices is to write an ALSA kernel driver that speaks HSS1394 with the devices and presents an ALSA MIDI device (with no data rate limits) for each to Linux applications. The advantage of doing this is that the devices will appear to all MIDI applications without them having to have specific support for HSS1394. This means they could be used in DAW applications as well. (FWIW, FFADO trunk has a program called test-scs that does exactly this, but using that is impractical because it depends on various internal parts of the FFADO source so would require bundling all of FFADO with the application, and it's unable to see non-audio devices due to FFADO's intended use (so it sees the SCS.1m but not the .1d.)) So my questions are: where can I find documentation on writing a new MIDI driver for ALSA? And is anyone else interested in helping me do this? Thank you very much for your time and work. Help me take the last step to my dream DJ setup! (Linux, FLAC and SCS.1 controllers.) Sincerely, Sean M. Pappalardo "D.J. Pegasus" Mixxx Developer - Controller Specialist

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[alsa-devel] Using UCM with PulseAudio
by Arun Raghavan 23 Jul '12

23 Jul '12

Hello, I've been working on porting PulseAudio to Android on the OMAP4-based Galaxy Nexus, and have recently looking at the policy-bits. The basic porting has been greatly simplified with Wei Feng's work to renew PulseAudio UCM integration and Liam's help with fixing up some of the UCM config for the Galaxy Nexus. I am, however, facing some trouble mapping how the hardware is used to how UCM presents it (or, perhaps, the UCM-PA mapping does). Some simplified background: the devices of interest on the OMAP4 SoC are the main hifi PCM which can be routed to various outputs, the modem PCM which is not used for actual output but to enable use of the modem during calls, and the tones PCM which is intended to be used for playing ringtones, aiui. The first problem is mutual exclusivity of verbs. From what I can understand, verbs are intended to be mutually exclusive -- if you have a HiFi verb and a VoiceCall verb, only one may be used at a time. We have mapped verbs to card profiles, which offer the same guarantee. However, on Android (which is a fair example of the kind of audio policy we might want), the HiFi verb PCMs maybe used while the VoiceCall PCMs are open. This is done, for example, to play an end-of-call tone from the CPU while the modem PCMs are still held open. Is there some way to do this with UCM? The second problem is having separate PCMs for modifiers. In the OMAP4 profile, ringtone playback is exposed via a PlayTone modifier which corresponds to a separate PCM from regular HiFi playback. In the UCM-PA mapping we decided on, modifiers were implemented as device intended roles on a sink, so that when a stream with that role came in, we could enable the modifier, and disable it when such a stream ends. However, this doesn't account for switching the PCM on which playback is occurring. Should we be creating a separate sink for such modifiers (with lower priority, so they're not routed to unless there's a stream with the required role coming in)? Or should we be reopening the PCM for this? Finally, a question also related to modifiers -- is it expected that there will never be a case where a stream that requires no modifier is being played while a stream that does require a modifier also exists? If not, what kind of policy should we have for enabling the modifier or not? Cheers, Arun

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[alsa-devel] [PATCH 0/2] RFC: support for audio wall clock
by Pierre-Louis Bossart 21 Jul '12

21 Jul '12

V2 of patches to show how the audio wall clock (eg. the HDAudio one) can be used to provide precise audio timestamps and track the drift between audio time and system time. On my laptop I am able to track a 7ppm delta. I will explain in more details how this can be used during LPC in San Diego, but with the summer coming I'd like to get feedback as there are changes to the core/alsa-lib that haven't been modified in years. TODO: - 64-bit alignment, verification of pcm_compat.c changes - handling of digital inputs, need to disable wall-clock timestamps in that case (feedback from Clemens) - check differences in protocol - change timecounter to avoid accumulation of rounding errors in cycle->ns conversions - check overflows - rework the timecounter init on trigger Pierre-Louis Bossart (2): ALSA: core: add hooks for audio timestamps read from WALLCLOCK ALSA: hda: support for wallclock timestamps include/sound/asound.h | 7 ++++- include/sound/pcm.h | 2 + sound/core/pcm_compat.c | 13 +++++++++- sound/core/pcm_lib.c | 15 ++++++++++- sound/core/pcm_native.c | 2 + sound/pci/hda/hda_intel.c | 54 ++++++++++++++++++++++++++++++++++++++++++++- 6 files changed, 86 insertions(+), 7 deletions(-) -- 1.7.6.5

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[alsa-devel] [PATCH] ALSA: hda - Add DeviceID for Haswell HDA
by Wang Xingchao 16 Jul '12

16 Jul '12

this patch add proper id for Haswell HDA Controller. Signed-off-by: Wang Xingchao <xingchao.wang(a)intel.com> --- sound/pci/hda/hda_intel.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 2b6392b..f530c40 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -151,6 +151,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6}," "{Intel, CPT}," "{Intel, PPT}," "{Intel, LPT}," + "{Intel, HPT}," "{Intel, PBG}," "{Intel, SCH}," "{ATI, SB450}," @@ -3253,6 +3254,10 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { { PCI_DEVICE(0x8086, 0x8c20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | AZX_DCAPS_BUFSIZE}, + /* Haswell */ + { PCI_DEVICE(0x8086, 0x0c0c), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | + AZX_DCAPS_BUFSIZE}, /* SCH */ { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | -- 1.7.9.5

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[alsa-devel] [PATCH] powerpc: select PPC_CLOCK unconditionally for FSL_SOC
by Shawn Guo 10 Jul '12

10 Jul '12

Freescale PowerPC SoCs share a number of IP blocks with Freescale ARM/IMX SoCs, FlexCAN, SSI, FEC, eSDHC, USB, etc. There are some effort consolidating those drivers to make them work for both architectures. One outstanding difference between two architectures is ARM/IMX will turn off module clocks during platform initialization for power saving and expects drivers manage clocks using clk API, while PowerPC mostly does not do that, and thus does not always build in clk API. Listing all those driver Kconfig options in "select PPC_CLOCK if" seems not scalable for long term maintenance, and could easily introduce Kconfig recursive dependency. This patch chooses to select PPC_CLOCK unconditionally for FSL_SOC to always build clk API for PowerPC in. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- arch/powerpc/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/arch/powerpc/Kconfig b/arch/powerpc/Kconfig index feab3ba..63fa7fb 100644 --- a/arch/powerpc/Kconfig +++ b/arch/powerpc/Kconfig @@ -661,7 +661,7 @@ config SBUS config FSL_SOC bool select HAVE_CAN_FLEXCAN if NET && CAN - select PPC_CLOCK if CAN_FLEXCAN + select PPC_CLOCK config FSL_PCI bool -- 1.7.5.4

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[alsa-devel] A problem with USB audio devices and the UHCI scheduler
by Monty Montgomery 04 Jul '12

04 Jul '12

A question for Clemens et al... Background: I can run a USB audio device at 6ch/24 bit single duplex, but I cannot run the same device at 16 bit stereo full duplex. UHCI and EHCI both say 'not enough bandwidth' (OHCI works). EHCI's scheduling issues are well known, but I was surprised UHCI could not schedule this either. The cause is simple; UHCI (and EHCI) use the endpoint's wMaxPacketSize for all bandwidth allocation. My device supports all of its sample rates on the same altset, with a wMaxPacketSize matching the highest (96kHz) rate. So, the HCD is reserving most of the USB bus's bandwidth regardless of the actual requirements. How should I approach fixing this in the driver? There's currently no means to communicate a desired bandwidth reservation to the HCD from USB audio, so I'd have to extend even the quirks. I was considering shadowing the endpoint's original wMaxPacketSize in the substream struct, and replacing the value in the endpoint's native descriptor while the stream is running. I have no idea how much trouble that would cause (eg, corruption elsewhere, or middle finger from Linus). Suggestions? For now I have things working by disabling UHCI's initial bandwidth checks (as there is in fact more than enough bandwidth to schedule what I'm doing) but I'd like a more permanent solution. Monty

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[alsa-devel] [PATCH] ASoC: tlv320aic3x: add deemphasis switch
by Prchal Jiří 04 Jul '12

04 Jul '12

This patch adds missing deemphasis switch. Signed-off-by: Jiri Prchal <jiri.prchal(a)aksignal.cz> --- sound/soc/codecs/tlv320aic3x.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 4ed6c20..2265e94 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -381,6 +381,9 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_ENUM("Left AGC Decay time", aic3x_agc_enum[LAGC_DEC_ENUM]), SOC_ENUM("Right AGC Decay time", aic3x_agc_enum[RAGC_DEC_ENUM]), + /* De-emphasis */ + SOC_DOUBLE("De-emphasis Switch", AIC3X_CODEC_DFILT_CTRL, 2, 0, 0x01, 0), + /* Input */ SOC_DOUBLE_R_TLV("PGA Capture Volume", LADC_VOL, RADC_VOL, 0, 119, 0, adc_tlv), -- 1.7.7

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[alsa-devel] [PATCH V3 0/4] Add support for STA529 codec, spdif_receiver and SPEAr spdif_out controller
by Rajeev Kumar 03 Jul '12

03 Jul '12

Earlier I had circulated Patch set V2. Please find below the link for the same. http://mailman.alsa-project.org/pipermail/alsa-devel/2012-June/052868.html. Some of the patches got applied and for rest I am sending Version V3. Patches that got applied(from V2 of patch set) are listed below: 1. [PATCH V2 3/9] sound:asoc: Add support for synopsys i2s controller as per ASoC framework. 2. [PATCH V2 4/9] sound:asoc: Add support for SPEAr ASoC pcm layer. 3. [PATCH V2 5/9] sound:asoc:spdif_in: Add spdif IN support Changes since V2: 1. In-corporated review comments received on V2 Rajeev Kumar (2): ASoC: STA529: Add support for STA529 Audio Codec ASoC: SPEAr: Add Kconfig and Makefile to support SPEAr audio driver Vipin Kumar (2): ASoC: spdif_receiver: Add support for spdif in Audio Codec ASoC: SPEAr spdif_out: Add spdif out support sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/spdif_receiver.c | 67 ++++++ sound/soc/codecs/sta529.c | 441 +++++++++++++++++++++++++++++++++++++ sound/soc/spear/Kconfig | 22 ++ sound/soc/spear/Makefile | 4 + sound/soc/spear/spdif_out.c | 387 ++++++++++++++++++++++++++++++++ sound/soc/spear/spdif_out_regs.h | 79 +++++++ 8 files changed, 1007 insertions(+), 1 deletions(-) create mode 100644 sound/soc/codecs/spdif_receiver.c create mode 100644 sound/soc/codecs/sta529.c create mode 100644 sound/soc/spear/Kconfig create mode 100644 sound/soc/spear/Makefile create mode 100644 sound/soc/spear/spdif_out.c create mode 100644 sound/soc/spear/spdif_out_regs.h

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[alsa-devel] [PATCH] ASoC: dapm: Fix snd_soc_dapm_put_volsw() connect
by Benoît Thébaudeau 03 Jul '12

03 Jul '12

snd_soc_dapm_put_volsw() sets connect incorrectly in the case max > 1 with invert. In that case, the raw disconnect value should be max, which corresponds to the userspace value 0. This use case currently does not appear upstream, but it could break SOC_DAPM_SINGLE() or SOC_DAPM_SINGLE_TLV() elsewhere or in the future. Cc: Liam Girdwood <lrg(a)ti.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: <alsa-devel(a)alsa-project.org> Signed-off-by: Benoît Thébaudeau <benoit.thebaudeau(a)advansee.com> --- .../sound/soc/soc-dapm.c | 8 +------- 1 file changed, 1 insertion(+), 7 deletions(-) diff --git linux-next-HEAD-6c86b58.orig/sound/soc/soc-dapm.c linux-next-HEAD-6c86b58/sound/soc/soc-dapm.c index 405841c..5ef082f 100644 --- linux-next-HEAD-6c86b58.orig/sound/soc/soc-dapm.c +++ linux-next-HEAD-6c86b58/sound/soc/soc-dapm.c @@ -2515,19 +2515,13 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, int wi; val = (ucontrol->value.integer.value[0] & mask); + connect = !!val; if (invert) val = max - val; mask = mask << shift; val = val << shift; - if (val) - /* new connection */ - connect = invert ? 0 : 1; - else - /* old connection must be powered down */ - connect = invert ? 1 : 0; - mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); change = snd_soc_test_bits(widget->codec, reg, mask, val);

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