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

[alsa-devel] [RFC][PATCH 00/17] Enhancement for firewire-lib
by Takashi Sakamoto 18 Dec '13

18 Dec '13

This series of patch extend functionalities of current firewire-lib: - support for both in/out AMDTP stream - support for duplex stream synchronization (blocking mode only) - support for both MIDI/PCM stream - support for both in/out CMP - support for some AV/C general commands Additionally, at the last of this series, some quirks for Fireworks are added. I'm happy to receive your comments, especially for sort processing. Takashi Sakamoto (17): firewire-lib: Rename functions, structure, member for both direction firewire-lib: Add macros instead of fixed value for AMDTP firewire-lib: Add 'direction' member to 'amdtp_stream' structure firewire-lib: Split some codes into functions to reuse in future firewire-lib: Add support for AMDTP transmit stream and PCM capture firewire-lib: Add support for MIDI capture/playback firewire-lib: give syt value to handle_out_packet() firewire-lib: Add support for duplex streams with synchronization firewire-lib: Add sort function for transmitted packet firewire-lib: Add transfer delay to synchronized duplex streams firewire-lib: Add support for channel mapping firewire-lib: Rename macros, variables and functions for CMP firewire-lib: Add 'direction' member to 'cmp_connection' structure firewire-lib: Add handling output connection by CMP firewire-lib: Add a new function to check others' connection firewire-lib: Add some AV/C general commands firewire-lib: Add quirks for Fireworks chipset sound/firewire/amdtp.c | 1014 +++++++++++++++++++++++++++++++++++---------- sound/firewire/amdtp.h | 161 +++++-- sound/firewire/cmp.c | 217 +++++++--- sound/firewire/cmp.h | 14 +- sound/firewire/dice.c | 46 +- sound/firewire/fcp.c | 149 +++++++ sound/firewire/fcp.h | 16 + sound/firewire/speakers.c | 94 ++--- 8 files changed, 1319 insertions(+), 392 deletions(-) -- 1.8.3.2

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[alsa-devel] [PATCH] ASoC: kirkwood: Fix the CPU DAI rates
by Jean-Francois Moine 17 Dec '13

17 Dec '13

This patch fixes the rates declared in the CPU DAI parameters: - SNDRV_PCM_RATE_KNOT and the discrete rates SNDRV_PCM_RATE_xxx should not be used with SNDRV_PCM_RATE_CONTINUOUS, - SNDRV_PCM_RATE_CONTINUOUS asks for rate_min and rate_max, - the device may do streaming down to 5512Hz. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/kirkwood/kirkwood-i2s.c | 24 +++++++++++----------- 1 file changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 0b18f65..3920a5e 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -473,17 +473,17 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk[2] = { .playback = { .channels_min = 1, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000 | - SNDRV_PCM_RATE_CONTINUOUS | - SNDRV_PCM_RATE_KNOT, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 192000, .formats = KIRKWOOD_I2S_FORMATS, }, .capture = { .channels_min = 1, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000 | - SNDRV_PCM_RATE_CONTINUOUS | - SNDRV_PCM_RATE_KNOT, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 192000, .formats = KIRKWOOD_I2S_FORMATS, }, .ops = &kirkwood_i2s_dai_ops, @@ -494,17 +494,17 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk[2] = { .playback = { .channels_min = 1, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000 | - SNDRV_PCM_RATE_CONTINUOUS | - SNDRV_PCM_RATE_KNOT, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 192000, .formats = KIRKWOOD_SPDIF_FORMATS, }, .capture = { .channels_min = 1, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000 | - SNDRV_PCM_RATE_CONTINUOUS | - SNDRV_PCM_RATE_KNOT, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 192000, .formats = KIRKWOOD_SPDIF_FORMATS, }, .ops = &kirkwood_i2s_dai_ops, -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

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[alsa-devel] While writing to speaker sometimes segmentation fault comes.
by ranjith vs 16 Dec '13

16 Dec '13

Hello, I am streaming audio from vlc,in the following formats sampling frequency 44100 stereo 16 bit.On the receiver side receiveing pcm frames and writing to speaker using alsa interface. But sometimes i'm getting segmentation fault in the snd_pcm_writei function. And one more thing is that eventhough I try to set the buffer size to 4096. It is setting it's own value.But i'm writing the buffer of length which I received from pcm_avail_upadte function. Thanks & Regards, Ranjith

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[alsa-devel] [PATCH v2] ASoC: SGTL5000: Fix kernel failed while trying to get optional VDDD supply.
by Xiubo Li 16 Dec '13

16 Dec '13

The SGTL5000 requires 2 external power supplies: VDDA and VDDIO. An optional third external power supply VDDD may be provided externally to achieve lower power.If an external supply is not used for VDDD, the SGTL5000 driver will register it's own regulator device, and then provides the VDDD supply consumer, and now there will be two regulator devices exist, local regulator for VDDD and platform regulator for VDDIO, VDDA. ******************* * *---------|3.3V VDDIO * * * SGTL5000 codec *---x VDDD * * * *---------|3.3V VDDA ******************* If an external supply is not used for VDDD, in the DT or architecture-specific file, only "VDDA-supply" and "VDDIO-supply" properties will be presented. This caused the following kernel failed while trying to get the external VDDD supply before trying to register it's own regulator device. sgtl5000 0-000a: Failed to get supply 'VDDD': -19 Here use regulator_get_optional() trying to look at the fact that whether the external VDDD supply is used or not. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 25 ++++++++++++++++++++++--- 1 file changed, 22 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 1f4093f..90d6d1e 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1298,6 +1298,21 @@ static int sgtl5000_replace_vddd_with_ldo(struct snd_soc_codec *codec) return 0; } +static int sgtl5000_external_vddd_used(struct snd_soc_codec *codec) +{ + struct regulator *consumer; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + + consumer = regulator_get_optional(codec->dev, + sgtl5000->supplies[VDDD].supply); + if (IS_ERR(consumer)) + return 0; + + regulator_put(consumer); + + return 1; +} + static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) { int reg; @@ -1310,11 +1325,15 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) for (i = 0; i < ARRAY_SIZE(sgtl5000->supplies); i++) sgtl5000->supplies[i].supply = supply_names[i]; - ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies), + if (sgtl5000_external_vddd_used(codec)) { + ret = regulator_bulk_get(codec->dev, + ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); - if (!ret) + if (ret) + return ret; + external_vddd = 1; - else { + } else { ret = sgtl5000_replace_vddd_with_ldo(codec); if (ret) return ret; -- 1.8.4

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[alsa-devel] [PATCH] ALSA: hda - limit mic boost on Asus UX31[A, E]
by Oleksij Rempel 15 Dec '13

15 Dec '13

This both devices need limit for internal dmic. Signed-off-by: Oleksij Rempel <linux(a)rempel-privat.de> --- sound/pci/hda/patch_realtek.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c4ad9d1..b617aa4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3798,6 +3798,7 @@ enum { ALC271_FIXUP_HP_GATE_MIC_JACK, ALC269_FIXUP_ACER_AC700, ALC269_FIXUP_LIMIT_INT_MIC_BOOST, + ALC269VB_FIXUP_LIMIT_INT_DMIC_BOOST, ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED, ALC269VB_FIXUP_ORDISSIMO_EVE2, ALC283_FIXUP_CHROME_BOOK, @@ -4075,6 +4076,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_THINKPAD_ACPI, }, + [ALC269VB_FIXUP_LIMIT_INT_DMIC_BOOST] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_limit_int_mic_boost, + .chained = true, + .chain_id = ALC269VB_FIXUP_DMIC, + }, [ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_limit_int_mic_boost, @@ -4189,8 +4196,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), - SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC), - SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_DMIC), + SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_LIMIT_INT_DMIC_BOOST), + SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_LIMIT_INT_DMIC_BOOST), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), -- 1.8.3.2

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[alsa-devel] [RFC][PATCH 00/13] bebob: a new driver for BridgeCo BeBoB based device
by Takashi Sakamoto 15 Dec '13

15 Dec '13

In previous series of patch, I showed enhancement of current firewire-lib. So I should show the way to use new functions in each driver side. This driver is under development but functionalities for streaming seems to be fixed. I'm happy to receive your comments. This is a new driver for BridgeCo BeBoB based device. FFADO (successor of FreeBoB project) currently give driver and mixer control in user space. My aim of this series of patch is just to give a support for kernel streaming. In theory, this driver just support kernel streaming and related functionality. But for my debugging, this module still includes some functionality which should be in user space. Additionally, there is a heavily customized version of BeBoB. For some reasons, this driver give a few functionality to control such device. Takashi Sakamoto (13): bebob: Add skelton for BridgeCo BeBoB based devices bebob: Add command and stream management bebob: Add proc interface for debug bebob: Add functionality to change sampling rate and control interface bebob: Add MIDI interface bebob: Add PCM interface bebob: Add hwdep interface to lock/unlock streaming bebob: Prepare for device specific operations bebob: add support for M-Audio Firewire series bebob: Add support for Focusrite Saffire/SaffirePro series bebob: Add support for Terratec PHASE series bebob: Add support for Yamaha GO series bebob: Add support for Presonus FIREBOX/FIREPOD include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 1 + sound/firewire/Kconfig | 29 ++ sound/firewire/Makefile | 1 + sound/firewire/bebob/Makefile | 6 + sound/firewire/bebob/bebob.c | 450 ++++++++++++++++ sound/firewire/bebob/bebob.h | 255 +++++++++ sound/firewire/bebob/bebob_command.c | 423 +++++++++++++++ sound/firewire/bebob/bebob_control.c | 331 ++++++++++++ sound/firewire/bebob/bebob_focusrite.c | 390 ++++++++++++++ sound/firewire/bebob/bebob_hwdep.c | 207 ++++++++ sound/firewire/bebob/bebob_maudio.c | 910 +++++++++++++++++++++++++++++++++ sound/firewire/bebob/bebob_midi.c | 182 +++++++ sound/firewire/bebob/bebob_pcm.c | 420 +++++++++++++++ sound/firewire/bebob/bebob_presonus.c | 75 +++ sound/firewire/bebob/bebob_proc.c | 164 ++++++ sound/firewire/bebob/bebob_stream.c | 724 ++++++++++++++++++++++++++ sound/firewire/bebob/bebob_terratec.c | 143 ++++++ sound/firewire/bebob/bebob_yamaha.c | 176 +++++++ 19 files changed, 4889 insertions(+), 1 deletion(-) create mode 100644 sound/firewire/bebob/Makefile create mode 100644 sound/firewire/bebob/bebob.c create mode 100644 sound/firewire/bebob/bebob.h create mode 100644 sound/firewire/bebob/bebob_command.c create mode 100644 sound/firewire/bebob/bebob_control.c create mode 100644 sound/firewire/bebob/bebob_focusrite.c create mode 100644 sound/firewire/bebob/bebob_hwdep.c create mode 100644 sound/firewire/bebob/bebob_maudio.c create mode 100644 sound/firewire/bebob/bebob_midi.c create mode 100644 sound/firewire/bebob/bebob_pcm.c create mode 100644 sound/firewire/bebob/bebob_presonus.c create mode 100644 sound/firewire/bebob/bebob_proc.c create mode 100644 sound/firewire/bebob/bebob_stream.c create mode 100644 sound/firewire/bebob/bebob_terratec.c create mode 100644 sound/firewire/bebob/bebob_yamaha.c -- 1.8.3.2

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[alsa-devel] how to create alsa nodes?
by loody 12 Dec '13

12 Dec '13

Dear all: In destop x86 system, the udev will automatically create the nodes in snd directory, such as pcmC0D0c, pcmC0D1c, etc. I try to let the usb sound device work on mips platform, but I have no idea what is the proper way to create these nodes. Should I 1. create them one by one by hand from copying the attributes in my PC (if I have 2 or more usb sound devices, I have to first plug them in pc and monitor the nodes) 2. smarter way to do so? -- Regards, miloody

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[alsa-devel] [PATCH 00/31] ARM: tegra: use common reset and DMA bindings
by Stephen Warren 12 Dec '13

12 Dec '13

From: Stephen Warren <swarren(a)nvidia.com> This series implements a common reset framework driver for Tegra, and updates all relevant Tegra drivers to use it. It also removes the custom DMA bindings and replaced them with the standard DMA DT bindings. Historically, the Tegra clock driver has exported a custom API for module reset. This series removes that API, and transitions DT and drivers to the new reset framework. The custom API used a "struct clk" to identify which module to reset, and consequently some DT bindings and drivers required clocks to be provided where they really needed just a reset identifier instead. Due to this known deficiency, I have always considered most Tegra bindings to be unstable. This series removes this excuse for instability, although I still consider some Tegra bindings unstable due to the need to convert to the common DMA bindings. Historically, Tegra DMA channels have been represented in DT using a custom nvidia,dma-request-selector property. Now that standard DMA DT bindings exist, convert all Tegra bindings, DTs, and drivers to use the standard instead. This series makes a DT-ABI-incompatible change to: - Require reset specifiers in DT where relevant. - Require standard DMA specifiers. - Remove clock specifiers from DT where they were only needed for reset. - Remove legacy DMA specifier properties. I anticipate merging this whole series into the Tegra and arm-soc trees as its own branch, due to internal dependencies. This branch will be stable and can then be merged into any other subsystem trees should any conflicts arise. This series depends on Peter's Tegra clock driver rework, available at git://nv-tegra.nvidia.com/user/pdeschrijver/linux tegra-clk-tegra124-0 (or whatever version of that gets included in 3.14) Cc: ac100(a)lists.launchpad.net Cc: Alan Stern <stern(a)rowland.harvard.edu> Cc: alsa-devel(a)alsa-project.org Cc: Bjorn Helgaas <bhelgaas(a)google.com> Cc: Dan Williams <dan.j.williams(a)intel.com> Cc: David Airlie <airlied(a)linux.ie> Cc: devel(a)driverdev.osuosl.org Cc: devicetree(a)vger.kernel.org Cc: Dmitry Torokhov <dmitry.torokhov(a)gmail.com> Cc: Dmitry Torokhov <dtor(a)mail.ru> Cc: dri-devel(a)lists.freedesktop.org Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: Ian Campbell <ijc+devicetree(a)hellion.org.uk> Cc: Julian Andres Klode <jak(a)jak-linux.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-i2c(a)vger.kernel.org Cc: linux-input(a)vger.kernel.org Cc: linux-pci(a)vger.kernel.org Cc: linux-serial(a)vger.kernel.org Cc: linux-spi(a)vger.kernel.org Cc: linux-tegra(a)vger.kernel.org Cc: linux-usb(a)vger.kernel.org Cc: Marc Dietrich <marvin24(a)gmx.de> Cc: Mark Brown <broonie(a)kernel.org> Cc: Mark Rutland <mark.rutland(a)arm.com> Cc: Mike Turquette <mturquette(a)linaro.org> Cc: Pawel Moll <pawel.moll(a)arm.com> Cc: pdeschrijver(a)nvidia.com Cc: Rob Herring <rob.herring(a)calxeda.com> Cc: Terje Bergström <tbergstrom(a)nvidia.com> Cc: treding(a)nvidia.com Cc: Wolfram Sang <wsa(a)the-dreams.de> Stephen Warren (31): ARM: tegra: add missing clock documentation to DT bindings ARM: tegra: document reset properties in DT bindings ARM: tegra: document use of standard DMA DT bindings ARM: tegra: update DT files to add reset properties ARM: tegra: update DT files to add DMA properties ARM: tegra: select the reset framework clk: tegra: implement a reset driver pci: tegra: use reset framework drm/tegra: use reset framework ARM: tegra: pass reset to tegra_powergate_sequence_power_up() dma: add channel request API that supports deferred probe dma: tegra: use reset framework dma: tegra: register as an OF DMA controller ASoC: dmaengine: support deferred probe for DMA channels ASoC: dmaengine: add custom DMA config to snd_dmaengine_pcm_config ASoC: tegra: use reset framework ASoC: tegra: call pm_runtime APIs around register accesses ASoC: tegra: allocate AHUB FIFO during probe() not startup() ASoC: tegra: convert to standard DMA DT bindings i2c: tegra: use reset framework staging: nvec: use reset framework spi: tegra: use reset framework spi: tegra: convert to standard DMA DT bindings serial: tegra: use reset framework serial: tegra: convert to standard DMA DT bindings Input: tegra-kbc - use reset framework USB: EHCI: tegra: use reset framework ARM: tegra: remove legacy clock entries from DT ARM: tegra: remove legacy DMA entries from DT clk: tegra: remove legacy reset APIs clk: tegra: remove bogus PCIE_XCLK .../bindings/arm/tegra/nvidia,tegra20-pmc.txt | 1 + .../bindings/clock/nvidia,tegra114-car.txt | 4 + .../bindings/clock/nvidia,tegra124-car.txt | 4 + .../bindings/clock/nvidia,tegra20-car.txt | 4 + .../bindings/clock/nvidia,tegra30-car.txt | 4 + .../devicetree/bindings/dma/tegra20-apbdma.txt | 9 ++ .../bindings/gpu/nvidia,tegra20-host1x.txt | 124 +++++++++++++++ .../devicetree/bindings/i2c/nvidia,tegra20-i2c.txt | 27 +++- .../bindings/input/nvidia,tegra20-kbc.txt | 9 ++ .../bindings/mmc/nvidia,tegra20-sdhci.txt | 9 ++ .../devicetree/bindings/nvec/nvidia,nvec.txt | 12 ++ .../bindings/pci/nvidia,tegra20-pcie.txt | 28 ++-- .../devicetree/bindings/pwm/nvidia,tegra20-pwm.txt | 9 ++ .../devicetree/bindings/rtc/nvidia,tegra20-rtc.txt | 3 + .../bindings/serial/nvidia,tegra20-hsuart.txt | 19 ++- .../bindings/sound/nvidia,tegra-audio-alc5632.txt | 7 +- .../bindings/sound/nvidia,tegra-audio-rt5640.txt | 7 +- .../bindings/sound/nvidia,tegra-audio-wm8753.txt | 7 +- .../bindings/sound/nvidia,tegra-audio-wm8903.txt | 7 +- .../bindings/sound/nvidia,tegra-audio-wm9712.txt | 7 +- .../bindings/sound/nvidia,tegra20-ac97.txt | 20 ++- .../bindings/sound/nvidia,tegra20-i2s.txt | 19 ++- .../bindings/sound/nvidia,tegra30-ahub.txt | 54 +++++-- .../bindings/sound/nvidia,tegra30-i2s.txt | 11 +- .../bindings/spi/nvidia,tegra114-spi.txt | 24 ++- .../bindings/spi/nvidia,tegra20-sflash.txt | 20 ++- .../bindings/spi/nvidia,tegra20-slink.txt | 20 ++- .../bindings/timer/nvidia,tegra20-timer.txt | 3 + .../bindings/timer/nvidia,tegra30-timer.txt | 3 + .../bindings/usb/nvidia,tegra20-ehci.txt | 7 +- arch/arm/boot/dts/tegra114.dtsi | 142 ++++++++++++++--- arch/arm/boot/dts/tegra20-paz00.dts | 2 + arch/arm/boot/dts/tegra20.dtsi | 132 ++++++++++++++-- arch/arm/boot/dts/tegra30.dtsi | 171 +++++++++++++++++---- arch/arm/mach-tegra/Kconfig | 2 + arch/arm/mach-tegra/powergate.c | 8 +- drivers/clk/tegra/clk-periph-gate.c | 22 --- drivers/clk/tegra/clk-periph.c | 40 ----- drivers/clk/tegra/clk-tegra114.c | 3 +- drivers/clk/tegra/clk-tegra124.c | 2 +- drivers/clk/tegra/clk-tegra20.c | 9 +- drivers/clk/tegra/clk-tegra30.c | 10 +- drivers/clk/tegra/clk.c | 55 ++++++- drivers/clk/tegra/clk.h | 3 +- drivers/dma/acpi-dma.c | 12 +- drivers/dma/dmaengine.c | 44 +++++- drivers/dma/of-dma.c | 12 +- drivers/dma/tegra20-apb-dma.c | 49 +++++- drivers/gpu/drm/tegra/Kconfig | 1 + drivers/gpu/drm/tegra/dc.c | 9 +- drivers/gpu/drm/tegra/drm.h | 3 + drivers/gpu/drm/tegra/gr3d.c | 22 ++- drivers/gpu/drm/tegra/hdmi.c | 14 +- drivers/i2c/busses/i2c-tegra.c | 13 +- drivers/input/keyboard/tegra-kbc.c | 13 +- drivers/pci/host/pci-tegra.c | 52 +++++-- drivers/spi/Kconfig | 3 + drivers/spi/spi-tegra114.c | 66 ++++---- drivers/spi/spi-tegra20-sflash.c | 18 ++- drivers/spi/spi-tegra20-slink.c | 66 ++++---- drivers/staging/nvec/nvec.c | 11 +- drivers/staging/nvec/nvec.h | 5 +- drivers/tty/serial/serial-tegra.c | 86 +++++------ drivers/usb/host/ehci-tegra.c | 14 +- include/dt-bindings/clock/tegra20-car.h | 2 +- include/dt-bindings/clock/tegra30-car.h | 2 +- include/linux/clk/tegra.h | 7 - include/linux/dmaengine.h | 7 + include/linux/of_dma.h | 9 +- include/linux/tegra-powergate.h | 4 +- include/sound/dmaengine_pcm.h | 6 + sound/soc/soc-generic-dmaengine-pcm.c | 82 +++++++--- sound/soc/tegra/Kconfig | 2 + sound/soc/tegra/tegra20_ac97.c | 11 -- sound/soc/tegra/tegra20_i2s.c | 20 +-- sound/soc/tegra/tegra30_ahub.c | 125 +++++++++------ sound/soc/tegra/tegra30_ahub.h | 11 +- sound/soc/tegra/tegra30_i2s.c | 97 ++++++------ sound/soc/tegra/tegra30_i2s.h | 3 + sound/soc/tegra/tegra_pcm.c | 17 +- sound/soc/tegra/tegra_pcm.h | 5 + 81 files changed, 1448 insertions(+), 558 deletions(-) -- 1.8.1.5

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[alsa-devel] [PATCHv8][ 1/5] ASoC: eukrea-tlv320: Add DT support.
by Denis Carikli 10 Dec '13

10 Dec '13

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 v7->v8: - fixed the vendor prefixes in the dts bindings. - updated the code and the documentation accordingly. - removed the need of an audio-codec property, since it's available in the ssi bindings. - the dai codec is now retrived trough the dt in order to support machines that have its codec on another i2c bus number. - driver compatible is now similar to the ones of the other drivers. ChangeLog v6->v7: - The cleanups went into another patch. - The support for fsl_ssi doesn't depend on dt anymore. - platform_name = "imx-ssi.0" is still set when in non-dt mode. --- .../devicetree/bindings/sound/eukrea-tlv320.txt | 21 ++++ sound/soc/fsl/Kconfig | 5 +- sound/soc/fsl/eukrea-tlv320.c | 108 +++++++++++++++++--- 3 files changed, 121 insertions(+), 13 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/eukrea-tlv320.txt diff --git a/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt b/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt new file mode 100644 index 0000000..0d7985c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt @@ -0,0 +1,21 @@ +Audio complex for Eukrea boards with tlv320aic23 codec. + +Required properties: +- compatible : "eukrea,asoc-tlv320" +- eukrea,model : The user-visible name of this sound complex. +- ssi-controller : The phandle of the SSI controller. +- fsl,mux-int-port : The internal port of the i.MX audio muxer (AUDMUX). +- fsl,mux-ext-port : The external port of the i.MX audio muxer. + +Note: The AUDMUX port numbering should start at 1, which is consistent with +hardware manual. + +Example: + + sound { + compatible = "eukrea,asoc-tlv320"; + eukrea,model = "imx51-eukrea-tlv320aic23"; + ssi-controller = <&ssi2>; + fsl,mux-int-port = <2>; + fsl,mux-ext-port = <3>; + }; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index b7ab71f..9c3cd64 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -161,12 +161,15 @@ config SND_SOC_EUKREA_TLV320 depends on MACH_EUKREA_MBIMX27_BASEBOARD \ || MACH_EUKREA_MBIMXSD25_BASEBOARD \ || MACH_EUKREA_MBIMXSD35_BASEBOARD \ - || MACH_EUKREA_MBIMXSD51_BASEBOARD + || MACH_EUKREA_MBIMXSD51_BASEBOARD \ + || OF depends on I2C select SND_SOC_TLV320AIC23 select SND_SOC_IMX_PCM_FIQ select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI + select SND_SOC_FSL_SSI + select SND_SOC_IMX_PCM_DMA help Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index 5983740..eb093d5 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -15,8 +15,11 @@ * */ +#include <linux/errno.h> #include <linux/module.h> #include <linux/moduleparam.h> +#include <linux/of.h> +#include <linux/of_platform.h> #include <linux/device.h> #include <linux/i2c.h> #include <sound/core.h> @@ -26,6 +29,7 @@ #include "../codecs/tlv320aic23.h" #include "imx-ssi.h" +#include "fsl_ssi.h" #include "imx-audmux.h" #define CODEC_CLOCK 12000000 @@ -41,7 +45,8 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret) { + /* fsl_ssi lacks the set_fmt ops. */ + if (ret && ret != -ENOTSUPP) { dev_err(cpu_dai->dev, "Failed to set the cpu dai format.\n"); return ret; @@ -63,11 +68,13 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, "Failed to set the codec sysclk.\n"); return ret; } + snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0); ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0, SND_SOC_CLOCK_IN); - if (ret) { + /* fsl_ssi lacks the set_sysclk ops */ + if (ret && ret != -EINVAL) { dev_err(cpu_dai->dev, "Can't set the IMX_SSP_SYS_CLK CPU system clock.\n"); return ret; @@ -84,14 +91,10 @@ static struct snd_soc_dai_link eukrea_tlv320_dai = { .name = "tlv320aic23", .stream_name = "TLV320AIC23", .codec_dai_name = "tlv320aic23-hifi", - .platform_name = "imx-ssi.0", - .codec_name = "tlv320aic23-codec.0-001a", - .cpu_dai_name = "imx-ssi.0", .ops = &eukrea_tlv320_snd_ops, }; static struct snd_soc_card eukrea_tlv320 = { - .name = "cpuimx-audio", .owner = THIS_MODULE, .dai_link = &eukrea_tlv320_dai, .num_links = 1, @@ -101,8 +104,65 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) { int ret; int int_port = 0, ext_port; + struct device_node *np = pdev->dev.of_node; + struct device_node *ssi_np, *codec_np; - if (machine_is_eukrea_cpuimx27()) { + eukrea_tlv320.dev = &pdev->dev; + if (np) { + ret = snd_soc_of_parse_card_name(&eukrea_tlv320, + "eukrea,model"); + if (ret) { + dev_err(&pdev->dev, + "eukrea,model node missing or invalid.\n"); + goto err; + } + + ssi_np = of_parse_phandle(pdev->dev.of_node, + "ssi-controller", 0); + if (!ssi_np) { + dev_err(&pdev->dev, + "ssi-controller missing or invalid.\n"); + ret = -ENODEV; + goto err; + } + + codec_np = of_parse_phandle(ssi_np, "codec-handle", 0); + if (codec_np) + eukrea_tlv320_dai.codec_of_node = codec_np; + else + dev_err(&pdev->dev, "codec-handle node missing or invalid.\n"); + + ret = of_property_read_u32(np, "fsl,mux-int-port", &int_port); + if (ret) { + dev_err(&pdev->dev, + "fsl,mux-int-port node missing or invalid.\n"); + return ret; + } + ret = of_property_read_u32(np, "fsl,mux-ext-port", &ext_port); + if (ret) { + dev_err(&pdev->dev, + "fsl,mux-ext-port node missing or invalid.\n"); + return ret; + } + + /* + * The port numbering in the hardware manual starts at 1, while + * the audmux API expects it starts at 0. + */ + int_port--; + ext_port--; + + eukrea_tlv320_dai.cpu_of_node = ssi_np; + eukrea_tlv320_dai.platform_of_node = ssi_np; + } else { + eukrea_tlv320_dai.cpu_dai_name = "imx-ssi.0"; + eukrea_tlv320_dai.platform_name = "imx-ssi.0"; + eukrea_tlv320_dai.codec_name = "tlv320aic23-codec.0-001a"; + eukrea_tlv320.name = "cpuimx-audio"; + } + + if (machine_is_eukrea_cpuimx27() || + of_find_compatible_node(NULL, NULL, "fsl,imx21-audmux")) { imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0, IMX_AUDMUX_V1_PCR_SYN | IMX_AUDMUX_V1_PCR_TFSDIR | @@ -119,8 +179,12 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) ); } else if (machine_is_eukrea_cpuimx25sd() || machine_is_eukrea_cpuimx35sd() || - machine_is_eukrea_cpuimx51sd()) { - ext_port = machine_is_eukrea_cpuimx25sd() ? 4 : 3; + machine_is_eukrea_cpuimx51sd() || + of_find_compatible_node(NULL, NULL, "fsl,imx31-audmux")) { + if (!np) + ext_port = machine_is_eukrea_cpuimx25sd() ? + 4 : 3; + imx_audmux_v2_configure_port(int_port, IMX_AUDMUX_V2_PTCR_SYN | IMX_AUDMUX_V2_PTCR_TFSDIR | @@ -134,14 +198,27 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) IMX_AUDMUX_V2_PDCR_RXDSEL(int_port) ); } else { - /* return happy. We might run on a totally different machine */ - return 0; + if (np) { + /* The eukrea,asoc-tlv320 driver was explicitely + * requested (through the device tree). + */ + dev_err(&pdev->dev, + "Missing or invalid audmux DT node.\n"); + return -ENODEV; + } else { + /* Return happy. + * We might run on a totally different machine. + */ + return 0; + } } - eukrea_tlv320.dev = &pdev->dev; ret = snd_soc_register_card(&eukrea_tlv320); +err: if (ret) dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + if (np) + of_node_put(ssi_np); return ret; } @@ -153,10 +230,17 @@ static int eukrea_tlv320_remove(struct platform_device *pdev) return 0; } +static const struct of_device_id imx_tlv320_dt_ids[] = { + { .compatible = "eukrea,asoc-tlv320"}, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_tlv320_dt_ids); + static struct platform_driver eukrea_tlv320_driver = { .driver = { .name = "eukrea_tlv320", .owner = THIS_MODULE, + .of_match_table = imx_tlv320_dt_ids, }, .probe = eukrea_tlv320_probe, .remove = eukrea_tlv320_remove, -- 1.7.9.5

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[alsa-devel] [PATCH 1/2] ASoC: samsung: Provide helper for DMA init
by Mark Brown 09 Dec '13

09 Dec '13

From: Mark Brown <broonie(a)linaro.org> In preparation for using the dmaengine helpers in ASoC rather than the dmaengine wrappers for the Samsung API wrap the configuration of dma_data. The dmaengine code expects different data to that used by the legacy API. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/samsung/ac97.c | 51 +++++++++++++++--------------------------------- sound/soc/samsung/dma.c | 8 ++++++++ sound/soc/samsung/dma.h | 3 +++ sound/soc/samsung/i2s.c | 2 +- sound/soc/samsung/pcm.c | 18 +++++++++-------- 5 files changed, 38 insertions(+), 44 deletions(-) diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 350ba23..4a88e36 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -221,24 +221,6 @@ static struct snd_ac97_bus_ops s3c_ac97_ops = { .reset = s3c_ac97_cold_reset, }; -static int s3c_ac97_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct s3c_dma_params *dma_data; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = &s3c_ac97_pcm_out; - else - dma_data = &s3c_ac97_pcm_in; - - snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); - - return 0; -} - static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -279,21 +261,6 @@ static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -static int s3c_ac97_hw_mic_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - return -ENODEV; - else - snd_soc_dai_set_dma_data(cpu_dai, substream, &s3c_ac97_mic_in); - - return 0; -} - static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -329,15 +296,27 @@ static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, } static const struct snd_soc_dai_ops s3c_ac97_dai_ops = { - .hw_params = s3c_ac97_hw_params, .trigger = s3c_ac97_trigger, }; static const struct snd_soc_dai_ops s3c_ac97_mic_dai_ops = { - .hw_params = s3c_ac97_hw_mic_params, .trigger = s3c_ac97_mic_trigger, }; +static int s3c_ac97_dai_probe(struct snd_soc_dai *dai) +{ + samsung_asoc_init_dma_data(dai, &s3c_ac97_pcm_out, &s3c_ac97_pcm_in); + + return 0; +} + +static int s3c_ac97_mic_dai_probe(struct snd_soc_dai *dai) +{ + samsung_asoc_init_dma_data(dai, NULL, &s3c_ac97_mic_in); + + return 0; +} + static struct snd_soc_dai_driver s3c_ac97_dai[] = { [S3C_AC97_DAI_PCM] = { .name = "samsung-ac97", @@ -354,6 +333,7 @@ static struct snd_soc_dai_driver s3c_ac97_dai[] = { .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .probe = s3c_ac97_dai_probe, .ops = &s3c_ac97_dai_ops, }, [S3C_AC97_DAI_MIC] = { @@ -365,6 +345,7 @@ static struct snd_soc_dai_driver s3c_ac97_dai[] = { .channels_max = 1, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .probe = s3c_ac97_mic_dai_probe, .ops = &s3c_ac97_mic_dai_ops, }, }; diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index fe2748b..ee23194 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -441,6 +441,14 @@ static struct snd_soc_platform_driver samsung_asoc_platform = { .pcm_free = dma_free_dma_buffers, }; +void samsung_asoc_init_dma_data(struct snd_soc_dai *dai, + struct s3c_dma_params *playback, + struct s3c_dma_params *capture) +{ + snd_soc_dai_init_dma_data(dai, playback, capture); +} +EXPORT_SYMBOL_GPL(samsung_asoc_init_dma_data); + int samsung_asoc_dma_platform_register(struct device *dev) { return snd_soc_register_platform(dev, &samsung_asoc_platform); diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h index 0e86315..fb09a1c 100644 --- a/sound/soc/samsung/dma.h +++ b/sound/soc/samsung/dma.h @@ -22,6 +22,9 @@ struct s3c_dma_params { char *ch_name; }; +void samsung_asoc_init_dma_data(struct snd_soc_dai *dai, + struct s3c_dma_params *playback, + struct s3c_dma_params *capture); int samsung_asoc_dma_platform_register(struct device *dev); void samsung_asoc_dma_platform_unregister(struct device *dev); diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index a5cbdb4..67d9fa9 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -963,7 +963,7 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) } clk_prepare_enable(i2s->clk); - snd_soc_dai_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); + samsung_asoc_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); if (other) { other->addr = i2s->addr; diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index e54256f..6a5e4bf 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -275,7 +275,6 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(rtd->cpu_dai); - struct s3c_dma_params *dma_data; void __iomem *regs = pcm->regs; struct clk *clk; int sclk_div, sync_div; @@ -284,13 +283,6 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, dev_dbg(pcm->dev, "Entered %s\n", __func__); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = pcm->dma_playback; - else - dma_data = pcm->dma_capture; - - snd_soc_dai_set_dma_data(rtd->cpu_dai, substream, dma_data); - /* Strictly check for sample size */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -461,10 +453,20 @@ static const struct snd_soc_dai_ops s3c_pcm_dai_ops = { .set_fmt = s3c_pcm_set_fmt, }; +static int s3c_pcm_dai_probe(struct snd_soc_dai *dai) +{ + struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, pcm->dma_playback, pcm->dma_capture); + + return 0; +} + #define S3C_PCM_RATES SNDRV_PCM_RATE_8000_96000 #define S3C_PCM_DAI_DECLARE \ .symmetric_rates = 1, \ + .probe = s3c_pcm_dai_probe, \ .ops = &s3c_pcm_dai_ops, \ .playback = { \ .channels_min = 2, \ -- 1.8.4.3

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