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

[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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[alsa-devel] [PATCH v2] ASoC: tlv320aic3x: add missing registers and bits
by Prchal Jiří 03 Jul '12

03 Jul '12

Second version of patch keeping formatting. Adds missing register default values to cache. Adds register and bits definitions in header file. Changes are for TLV320AIC310x based on data sheet. Signed-off-by: Jiri Prchal <jiri.prchal(a)aksignal.cz> --- /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.c.orig +++ /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.c.new @@ -122,7 +122,9 @@ 0x00, 0x00, 0x00, 0x00, /* 88 */ 0x00, 0x00, 0x00, 0x00, /* 92 */ 0x00, 0x00, 0x00, 0x00, /* 96 */ - 0x00, 0x00, 0x02, /* 100 */ + 0x00, 0x00, 0x02, 0x00, /* 100 */ + 0x00, 0x00, 0x00, 0x00, /* 104 */ + 0x00, 0x00, /* 108 */ }; --- /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.h +++ /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.h.new @@ -13,7 +13,7 @@ #define _AIC3X_H /* AIC3X register space */ -#define AIC3X_CACHEREGNUM 103 +#define AIC3X_CACHEREGNUM 110 /* Page select register */ #define AIC3X_PAGE_SELECT 0 @@ -74,6 +74,8 @@ #define HPLCOM_CFG 37 /* Right High Power Output control registers */ #define HPRCOM_CFG 38 +/* High Power Output Stage Control Register */ +#define HPOUT_SC 40 /* DAC Output Switching control registers */ #define DAC_LINE_MUX 41 /* High Power Output Driver Pop Reduction registers */ @@ -148,6 +150,17 @@ #define AIC3X_GPIOB_REG 101 /* Clock generation control register */ #define AIC3X_CLKGEN_CTRL_REG 102 +/* New AGC registers */ +#define LAGCN_ATTACK 103 +#define LAGCN_DECAY 104 +#define RAGCN_ATTACK 105 +#define RAGCN_DECAY 106 +/* New Programmable ADC Digital Path and I2C Bus Condition Register */ +#define NEW_ADC_DIGITALPATH 107 +/* Passive Analog Signal Bypass Selection During Powerdown Register */ +#define PASSIVE_BYPASS 108 +/* DAC Quiescent Current Adjustment Register */ +#define DAC_ICC_ADJ 109 /* Page select register bits */ #define PAGE0_SELECT 0 @@ -163,6 +176,10 @@ #define DUAL_RATE_MODE ((1 << 5) | (1 << 6)) #define LDAC2LCH (0x1 << 3) #define RDAC2RCH (0x1 << 1) +#define LDAC2RCH (0x2 << 3) +#define RDAC2LCH (0x2 << 1) +#define LDAC2MONOMIX (0x3 << 3) +#define RDAC2MONOMIX (0x3 << 1) /* PLL registers bitfields */ #define PLLP_SHIFT 0

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