April 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/5 v4] Add I2S/ADV7511 audio support for ARC AXS10x boards
by Jose Abreu 11 Apr '16

11 Apr '16

ARC AXS10x platforms consist of a mainboard with several peripherals. One of those peripherals is an HDMI output port controlled by the ADV7511 transmitter. This patch set adds audio for the ADV7511 transmitter and I2S audio for the AXS10x platform. Changes v3 -> v4: * Reintroduced custom PCM driver (see note below) * Use DT boolean to switch between ALSA DMA engine PCM or custom PCM * Use fifo depth to program I2S FCR * Update I2S documentation Changes v2 -> v3: * Removed pll_config functions (as suggested by Alexey Brodkin) * Removed HDMI start at adv7511_core (as suggested by Archit Taneja) * Use NOP functions for adv7511_audio (as suggested by Archit Taneja) * Added adv7511_audio_exit() function (as suggested by Archit Taneja) * Moved adv7511 to its own folder (as suggested by Archit Taneja) * Separated file rename of adv7511_core (as suggested by Emil Velikov) * Compile adv7511 as module if ALSA SoC is compiled as module * Load adv7511 audio only if declared in device tree (as suggested by Laurent Pinchart) * Dropped custom platform driver, using now ALSA DMA engine * Dropped IRQ handler for I2S Changes v1 -> v2: * DT bindings moved to separate patch (as suggested by Alexey Brodkin) * Removed defconfigs entries (as suggested by Alexey Brodkin) NOTE: Although the mainline I2S driver uses ALSA DMA engine, this controller can be built without DMA support so it was necessary to add this custom platform driver so that HDMI audio works in AXS boards. Jose Abreu (5): drm/i2c/adv7511: Rename and move to separate folder drm/i2c/adv7511: Add audio support ASoC: dwc: Use fifo depth to program FCR ASoC: dwc: Add custom PCM driver ASoC: dwc: Update DOCUMENTATION for I2S Driver .../bindings/display/bridge/adi,adv7511.txt | 3 + .../devicetree/bindings/sound/designware-i2s.txt | 5 + drivers/gpu/drm/i2c/Kconfig | 6 +- drivers/gpu/drm/i2c/Makefile | 2 +- drivers/gpu/drm/i2c/adv7511/Kconfig | 18 ++ drivers/gpu/drm/i2c/adv7511/Makefile | 3 + drivers/gpu/drm/i2c/{ => adv7511}/adv7511.h | 53 ++++ drivers/gpu/drm/i2c/adv7511/adv7511_audio.c | 310 +++++++++++++++++++++ .../drm/i2c/{adv7511.c => adv7511/adv7511_core.c} | 43 +-- include/sound/soc-dai.h | 1 + sound/soc/dwc/Kconfig | 9 + sound/soc/dwc/Makefile | 1 + sound/soc/dwc/designware.h | 70 +++++ sound/soc/dwc/designware_i2s.c | 106 +++++-- sound/soc/dwc/designware_pcm.c | 230 +++++++++++++++ 15 files changed, 796 insertions(+), 64 deletions(-) create mode 100644 drivers/gpu/drm/i2c/adv7511/Kconfig create mode 100644 drivers/gpu/drm/i2c/adv7511/Makefile rename drivers/gpu/drm/i2c/{ => adv7511}/adv7511.h (90%) create mode 100644 drivers/gpu/drm/i2c/adv7511/adv7511_audio.c rename drivers/gpu/drm/i2c/{adv7511.c => adv7511/adv7511_core.c} (97%) create mode 100644 sound/soc/dwc/designware.h create mode 100644 sound/soc/dwc/designware_pcm.c -- 1.9.1

3 21

[alsa-devel] [PATCH] ALSA: hda/realtek - Enable the ALC292 dock fixup on the Thinkpad T460s
by Sven Eckelmann 11 Apr '16

11 Apr '16

The Lenovo Thinkpad T460s requires the alc_fixup_tpt440_dock as well in order to get working sound output on the docking stations headphone jack. Patch tested on a Thinkpad T460s (20F9CT01WW) using a ThinkPad Ultradock on kernel 4.4.6. Signed-off-by: Sven Eckelmann <sven(a)narfation.org> Tested-by: Simon Wunderlich <sw(a)simonwunderlich.de> --- v3: - Removed ALC292_FIXUP_TPT460_DOCK v2: - rebased on current master sound/pci/hda/patch_realtek.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fefe83f..1402ba9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4760,6 +4760,7 @@ enum { ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC280_FIXUP_HP_HEADSET_MIC, ALC221_FIXUP_HP_FRONT_MIC, + ALC292_FIXUP_TPT460, }; static const struct hda_fixup alc269_fixups[] = { @@ -5409,6 +5410,12 @@ static const struct hda_fixup alc269_fixups[] = { { } }, }, + [ALC292_FIXUP_TPT460] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_tpt440_dock, + .chained = true, + .chain_id = ALC293_FIXUP_LENOVO_SPK_NOISE, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -5563,7 +5570,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2218, "Thinkpad X1 Carbon 2nd", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2223, "ThinkPad T550", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK), - SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE), + SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC292_FIXUP_TPT460), SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), @@ -5658,6 +5665,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC283_FIXUP_SENSE_COMBO_JACK, .name = "alc283-sense-combo"}, {.id = ALC292_FIXUP_TPT440_DOCK, .name = "tpt440-dock"}, {.id = ALC292_FIXUP_TPT440, .name = "tpt440"}, + {.id = ALC292_FIXUP_TPT460, .name = "tpt460"}, {} }; #define ALC225_STANDARD_PINS \

2 1

[alsa-devel] [PATCH v2] ALSA: hda/realtek - Enable the ALC292 dock fixup on the Thinkpad T460s
by Sven Eckelmann 11 Apr '16

11 Apr '16

The Lenovo Thinkpad T460s requires the alc_fixup_tpt440_dock as well in order to get working sound output on the docking stations headphone jack. Patch tested on a Thinkpad T460s (20F9CT01WW) using a ThinkPad Ultradock on kernel 4.4.6. Signed-off-by: Sven Eckelmann <sven(a)narfation.org> Tested-by: Simon Wunderlich <sw(a)simonwunderlich.de> --- v2: - rebased on current master sound/pci/hda/patch_realtek.c | 18 +++++++++++++++++- 1 file changed, 17 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fefe83f..36d371c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4760,6 +4760,8 @@ enum { ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC280_FIXUP_HP_HEADSET_MIC, ALC221_FIXUP_HP_FRONT_MIC, + ALC292_FIXUP_TPT460_DOCK, + ALC292_FIXUP_TPT460, }; static const struct hda_fixup alc269_fixups[] = { @@ -5409,6 +5411,18 @@ static const struct hda_fixup alc269_fixups[] = { { } }, }, + [ALC292_FIXUP_TPT460_DOCK] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_tpt440_dock, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI + }, + [ALC292_FIXUP_TPT460] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_disable_aamix, + .chained = true, + .chain_id = ALC292_FIXUP_TPT460_DOCK, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -5563,7 +5577,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2218, "Thinkpad X1 Carbon 2nd", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2223, "ThinkPad T550", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK), - SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE), + SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC292_FIXUP_TPT460), SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), @@ -5658,6 +5672,8 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC283_FIXUP_SENSE_COMBO_JACK, .name = "alc283-sense-combo"}, {.id = ALC292_FIXUP_TPT440_DOCK, .name = "tpt440-dock"}, {.id = ALC292_FIXUP_TPT440, .name = "tpt440"}, + {.id = ALC292_FIXUP_TPT460_DOCK, .name = "tpt460-dock"}, + {.id = ALC292_FIXUP_TPT460, .name = "tpt460"}, {} }; #define ALC225_STANDARD_PINS \ -- 2.8.0.rc3

2 3

[alsa-devel] A tweak to the intel codec to get rid of the hiss.
by Cyrille Dunant 11 Apr '16

11 Apr '16

I apologise for the noise on the list, but I noted that on a number of HDA card, the default produces a small static noise, and I tracked it down to some channels which I believe ought to be muted. Attached is the output of alsa-info.sh, and the diff from the python hda tweak tool. This helped both for my laptop and desktop (both HDA cards, different models). Cheers! Diff for codec 1/0 (0x10ec0280): --- +++ @@ -17,17 +17,17 @@ IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[4]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x41d: Stereo Amp-Out Device: name="ALC3220 Analog", type="Audio", device=0 Control: name="PCM Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=1, idx=0, ofs=0 Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 - Amp-Out vals: [0x2e 0x2e] + Amp-Out vals: [0x2f 0x2f] Converter: stream=8, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x03 [Audio Output] wcaps 0x41d: Stereo Amp-Out Control: name="Headphone Playback Volume", index=0, device=0 @@ -66,17 +66,17 @@ rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x3f, stepsize=0x02, mute=1 - Amp-In vals: [0x97 0x97] + Amp-In vals: [0x17 0x17] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Connection: 1 @@ -87,27 +87,27 @@ ControlAmp: chs=3, dir=1, idx=2, ofs=0 Control: name="Front Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=1, idx=2, ofs=0 Control: name="Rear Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=1, idx=0, ofs=0 Control: name="Rear Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=1, idx=0, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 - Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] + Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Connection: 5 0x18 0x19 0x1a 0x1b 0x1d Node 0x0c [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 - Amp-In vals: [0x00 0x00] [0x00 0x00] + Amp-In vals: [0x00 0x00] [0x80 0x80] Connection: 2 0x03 0x0b Node 0x0e [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0f [Audio Mixer] wcaps 0x20010a: Mono Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: 0x80 0x00 Connection: 2 0x02 0x0

2 1

[alsa-devel] [PATCH 0/6] ASoC: davinci-mcbsp: add binding for McBSP
by Petr Kulhavy 11 Apr '16

11 Apr '16

This series of patches adds devicetree support for the davinci McBSP audio interface.

3 20

[alsa-devel] Checkpatch warnings on switch case fallthru
by Harvey, Ryan 11 Apr '16

11 Apr '16

Hello, I receive a set of checkpatch warnings against my switch fallthru structure in readable_registers(). I¹m able to eliminate these warnings with a preceding comment, but that does messy the code. Is there a method better than switch for accomplishing this? static bool cs43130_readable_register(struct device *dev, unsigned int reg) { switch (reg) { case CS43130_DEVID_AB ... CS43130_SYS_CLK_CTL_1: case CS43130_PWDN_CTL: case CS43130_PLL_SET_1 ... CS43130_PLL_SET_6: case CS43130_PLL_SET_7: case CS43130_PLL_SET_8: case CS43130_PLL_SET_9: case CS43130_PLL_SET_10: case CS43130_CLKOUT_CTL: case CS43130_ASP_NUM_1 ... CS43130_ASP_FRAME_CONF: case CS43130_HP_OUT_CTL_1: case CS43130_PCM_FILT_OPT ... CS43130_PCM_PATH_CTL_2: case CS43130_CLASS_H_CTL: case CS43130_HP_DETECT ... CS43130_HP_STATUS: case CS43130_HP_LOAD_1: case CS43130_HP_LOAD_STAT: case CS43130_INT_MASK_1 ... CS43130_INT_MASK_5: return true; default: return false; } } 140: [WARNING] Possible switch case/default not preceeded by break or fallthrough comment + case CS43130_PLL_SET_7: 141: [WARNING] Possible switch case/default not preceeded by break or fallthrough comment + case CS43130_PLL_SET_8: 142: [WARNING] Possible switch case/default not preceeded by break or fallthrough comment + case CS43130_PLL_SET_9: 143: [WARNING] Possible switch case/default not preceeded by break or fallthrough comment + case CS43130_PLL_SET_10: 144: [WARNING] Possible switch case/default not preceeded by break or fallthrough comment + case CS43130_CLKOUT_CTL: 152: [WARNING] Possible switch case/default not preceeded by break or fallthrough comment + case CS43130_HP_OUT_CTL_1: 154: [WARNING] Possible switch case/default not preceeded by break or fallthrough comment + case CS43130_CLASS_H_CTL: 156: [WARNING] Possible switch case/default not preceeded by break or fallthrough comment + case CS43130_HP_LOAD_1: 160: [WARNING] Possible switch case/default not preceeded by break or fallthrough comment + case CS43130_HP_LOAD_STAT: Thanks -Ryan

4 5

[alsa-devel] [PATCH] ASoC: rockchip: i2s: configure the sdio pins' iomux mode
by Sugar Zhang 11 Apr '16

11 Apr '16

There are 3 i2s sdio pins, which iomux mode is as follows: - sdi3_sdo1 - sdi2_sdo2 - sdi1_sdo3 we need to configure these pins' iomux mode via the GRF register when use multi channel playback/capture. Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> --- .../devicetree/bindings/sound/rockchip-i2s.txt | 5 +++ sound/soc/rockchip/rockchip_i2s.c | 39 +++++++++++++++++++++- sound/soc/rockchip/rockchip_i2s.h | 8 +++++ 3 files changed, 51 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt index 6e86d8a..ad72a7d 100644 --- a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt +++ b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt @@ -23,6 +23,11 @@ Required properties: - rockchip,playback-channels: max playback channels, if not set, 8 channels default. - rockchip,capture-channels: max capture channels, if not set, 2 channels default. +Required properties for controller which support multi channels playback/capture: + +- rockchip,grf: Should be phandle/offset pair. the phandle of the syscon node for GRF register, + and the offset of the GRF for control register. + Example for rk3288 I2S controller: i2s@ff890000 { diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 2f8e204..bc72780 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -11,6 +11,7 @@ */ #include <linux/module.h> +#include <linux/mfd/syscon.h> #include <linux/delay.h> #include <linux/of_gpio.h> #include <linux/clk.h> @@ -33,6 +34,8 @@ struct rk_i2s_dev { struct snd_dmaengine_dai_dma_data playback_dma_data; struct regmap *regmap; + struct regmap *grf; + u32 iocfg_reg; bool is_master_mode; }; @@ -277,6 +280,29 @@ static int rockchip_i2s_hw_params(struct snd_pcm_substream *substream, I2S_TXCR_VDW_MASK | I2S_TXCR_CSR_MASK, val); + if (!IS_ERR(i2s->grf)) { + regmap_read(i2s->regmap, I2S_TXCR, &val); + val &= I2S_TXCR_CSR_MASK; + + switch (val) { + case I2S_CHN_4: + val = I2S_IO_4CH_OUT_6CH_IN; + break; + case I2S_CHN_6: + val = I2S_IO_6CH_OUT_4CH_IN; + break; + case I2S_CHN_8: + val = I2S_IO_8CH_OUT_2CH_IN; + break; + default: + val = I2S_IO_2CH_OUT_8CH_IN; + break; + } + + regmap_write(i2s->grf, i2s->iocfg_reg, + I2S_IO_DIRECTION_MASK << 16 | val); + } + regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_TDL_MASK, I2S_DMACR_TDL(16)); regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_RDL_MASK, @@ -478,6 +504,18 @@ static int rockchip_i2s_probe(struct platform_device *pdev) return -ENOMEM; } + i2s->dev = &pdev->dev; + + i2s->grf = syscon_regmap_lookup_by_phandle(node, "rockchip,grf"); + if (!IS_ERR(i2s->grf)) { + ret = of_property_read_u32_index(node, "rockchip,grf", + 1, &i2s->iocfg_reg); + if (ret) { + dev_err(&pdev->dev, "Can't get iocfg_reg offset\n"); + return ret; + } + } + /* try to prepare related clocks */ i2s->hclk = devm_clk_get(&pdev->dev, "i2s_hclk"); if (IS_ERR(i2s->hclk)) { @@ -517,7 +555,6 @@ static int rockchip_i2s_probe(struct platform_device *pdev) i2s->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; i2s->capture_dma_data.maxburst = 4; - i2s->dev = &pdev->dev; dev_set_drvdata(&pdev->dev, i2s); pm_runtime_enable(&pdev->dev); diff --git a/sound/soc/rockchip/rockchip_i2s.h b/sound/soc/rockchip/rockchip_i2s.h index dc6e2c7..9a6aabf 100644 --- a/sound/soc/rockchip/rockchip_i2s.h +++ b/sound/soc/rockchip/rockchip_i2s.h @@ -236,4 +236,12 @@ enum { #define I2S_TXDR (0x0024) #define I2S_RXDR (0x0028) +/* io direction cfg register */ +#define I2S_IO_DIRECTION_SHIFT 11 +#define I2S_IO_DIRECTION_MASK (7 << I2S_IO_DIRECTION_SHIFT) +#define I2S_IO_8CH_OUT_2CH_IN (0 << I2S_IO_DIRECTION_SHIFT) +#define I2S_IO_6CH_OUT_4CH_IN (1 << I2S_IO_DIRECTION_SHIFT) +#define I2S_IO_4CH_OUT_6CH_IN (3 << I2S_IO_DIRECTION_SHIFT) +#define I2S_IO_2CH_OUT_8CH_IN (7 << I2S_IO_DIRECTION_SHIFT) + #endif /* _ROCKCHIP_IIS_H */ -- 1.9.1

4 4

[alsa-devel] [PATCH] ASoC: Add max98371 codec driver
by anish kumar 10 Apr '16

10 Apr '16

Signed-off-by: anish kumar <yesanishhere(a)gmail.com> --- Changes in this version: 1. Removed the probe callback as it was not used. 2. Replaced the control with tlv .../devicetree/bindings/sound/max98371.txt | 17 + sound/soc/codecs/max98371.c | 427 +++++++++++++++++++++ sound/soc/codecs/max98371.h | 64 +++ 3 files changed, 508 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98371.txt create mode 100644 sound/soc/codecs/max98371.c create mode 100644 sound/soc/codecs/max98371.h diff --git a/Documentation/devicetree/bindings/sound/max98371.txt b/Documentation/devicetree/bindings/sound/max98371.txt new file mode 100644 index 0000000..6c28523 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max98371.txt @@ -0,0 +1,17 @@ +max98371 codec + +This device supports I2C mode only. + +Required properties: + +- compatible : "maxim,max98371" +- reg : The chip select number on the I2C bus + +Example: + +&i2c { + max98371: max98371@0x31 { + compatible = "maxim,max98371"; + reg = <0x31>; + }; +}; diff --git a/sound/soc/codecs/max98371.c b/sound/soc/codecs/max98371.c new file mode 100644 index 0000000..fe7f08f --- /dev/null +++ b/sound/soc/codecs/max98371.c @@ -0,0 +1,427 @@ +/* + * max98371.c -- ALSA SoC Stereo MAX98371 driver + * + * Copyright 2015-16 Maxim Integrated Products + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "max98371.h" + +static const char *const monomix_text[] = { + "Left", "Right", "LeftRightDiv2", +}; + +static const char *const voltage_level_text[] = { + "9.5dB", "10.5dB", "11.5dB", "11.5dB", "12.5dB", + "13.5dB", "14.5dB", "15.5dB", "16.5dB", "17.5dB", + "18.5dB", "19.5dB", "20.5dB" +}; + +static const char *const hpf_cutoff_txt[] = { + "Disable", "DC Block", "50Hz", + "100Hz", "200Hz", "400Hz", "800Hz", +}; + +static SOC_ENUM_SINGLE_DECL(max98371_voltage, MAX98371_GAIN, 0, + voltage_level_text); + +static SOC_ENUM_SINGLE_DECL(max98371_monomix, MAX98371_MONOMIX_CFG, 0, + monomix_text); + +static SOC_ENUM_SINGLE_DECL(max98371_hpf_cutoff, MAX98371_HPF, 0, + hpf_cutoff_txt); + +static struct reg_default max98371_reg[] = { + { 0x01, 0x00 }, + { 0x02, 0x00 }, + { 0x03, 0x00 }, + { 0x04, 0x00 }, + { 0x05, 0x00 }, + { 0x06, 0x00 }, + { 0x07, 0x00 }, + { 0x08, 0x00 }, + { 0x09, 0x00 }, + { 0x0A, 0x00 }, + { 0x10, 0x06 }, + { 0x11, 0x08 }, + { 0x14, 0x80 }, + { 0x15, 0x00 }, + { 0x16, 0x00 }, + { 0x18, 0x00 }, + { 0x19, 0x00 }, + { 0x1C, 0x00 }, + { 0x1D, 0x00 }, + { 0x1E, 0x00 }, + { 0x1F, 0x00 }, + { 0x20, 0x00 }, + { 0x21, 0x00 }, + { 0x22, 0x00 }, + { 0x23, 0x00 }, + { 0x24, 0x00 }, + { 0x25, 0x00 }, + { 0x26, 0x00 }, + { 0x27, 0x00 }, + { 0x28, 0x00 }, + { 0x29, 0x00 }, + { 0x2A, 0x00 }, + { 0x2B, 0x00 }, + { 0x2C, 0x00 }, + { 0x2D, 0x00 }, + { 0x2E, 0x0B }, + { 0x31, 0x00 }, + { 0x32, 0x18 }, + { 0x33, 0x00 }, + { 0x34, 0x00 }, + { 0x36, 0x00 }, + { 0x37, 0x00 }, + { 0x38, 0x00 }, + { 0x39, 0x00 }, + { 0x3A, 0x00 }, + { 0x3B, 0x00 }, + { 0x3B, 0x00 }, + { 0x3C, 0x00 }, + { 0x3D, 0x00 }, + { 0x3E, 0x00 }, + { 0x3F, 0x00 }, + { 0x40, 0x00 }, + { 0x41, 0x00 }, + { 0x42, 0x00 }, + { 0x43, 0x00 }, + { 0x4A, 0x00 }, + { 0x4B, 0x00 }, + { 0x4C, 0x00 }, + { 0x4D, 0x00 }, + { 0x4E, 0x00 }, + { 0x50, 0x00 }, + { 0x51, 0x00 }, + { 0x55, 0x00 }, + { 0x58, 0x00 }, + { 0x59, 0x00 }, + { 0x5C, 0x00 }, + { 0xFF, 0x43 }, +}; + +static bool max98371_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98371_IRQ_CLEAR1: + case MAX98371_IRQ_CLEAR2: + case MAX98371_IRQ_CLEAR3: + case MAX98371_VERSION: + return true; + default: + return false; + } +} + +static bool max98371_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98371_SOFT_RESET: + return false; + default: + return true; + } +}; + +static const unsigned int max98371_gain_tlv[] = { + 0, 8, TLV_DB_SCALE_ITEM(0, 50, 0), + 9, 10, TLV_DB_SCALE_ITEM(500, 100, 0), + 11, 15, TLV_DB_SCALE_ITEM(0, 0, 0), +}; + +static const unsigned int max98371_noload_gain_tlv[] = { + 0, 11, TLV_DB_SCALE_ITEM(0, 100, 0), +}; + +static const DECLARE_TLV_DB_SCALE(digital_tlv, -6300, 50, 1); + +static const struct snd_kcontrol_new max98371_snd_controls[] = { + SOC_SINGLE_TLV("Speaker Volume", MAX98371_GAIN, + MAX98371_GAIN_SHIFT, (1<<MAX98371_GAIN_WIDTH)-1, 0, + max98371_gain_tlv), + SOC_SINGLE_TLV("Digital Volume", MAX98371_DIGITAL_GAIN, 0, + (1<<MAX98371_DIGITAL_GAIN_WIDTH)-1, 1, digital_tlv), + SOC_SINGLE_TLV("Speaker No-load Volume", MAX98371_GAIN, + 0, (1<<MAX98371_NO_LOAD_WIDTH)-1, 0, + max98371_noload_gain_tlv), + SOC_SINGLE("DHT Attack Step", MAX98371_DHT, MAX98371_DHT_STEP, 3, 0), + SOC_SINGLE("DHT Attack Rate", MAX98371_DHT, 0, 7, 0), + SOC_ENUM("Voltage Level", max98371_voltage), + SOC_ENUM("Monomix Select", max98371_monomix), + SOC_ENUM("HPF Cutoff", max98371_hpf_cutoff), +}; + +static int max98371_dai_set_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max98371_priv *max98371 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + dev_err(codec->dev, "DAI clock mode unsupported"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + val |= 0; + break; + case SND_SOC_DAIFMT_RIGHT_J: + val |= MAX98371_DAI_RIGHT; + break; + case SND_SOC_DAIFMT_LEFT_J: + val |= MAX98371_DAI_LEFT; + break; + default: + dev_err(codec->dev, "DAI wrong mode unsupported"); + return -EINVAL; + } + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MODE_MASK, val); + return 0; +} + +static int max98371_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98371_priv *max98371 = snd_soc_codec_get_drvdata(codec); + int blr_clk_ratio, ch_size, channels = params_channels(params); + int rate = params_rate(params); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_16); + ch_size = 8; + break; + case SNDRV_PCM_FORMAT_S16_LE: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_16); + ch_size = 16; + break; + case SNDRV_PCM_FORMAT_S24_LE: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_32); + ch_size = 24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_32); + ch_size = 32; + break; + default: + pr_info("%s: format unsupported %d", + __func__, params_format(params)); + return -EINVAL; + } + + /* BCLK/LRCLK ratio calculation */ + blr_clk_ratio = channels * ch_size; + switch (blr_clk_ratio) { + case 32: + regmap_update_bits(max98371->regmap, + MAX98371_DAI_CLK, + MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_32); + break; + case 48: + regmap_update_bits(max98371->regmap, + MAX98371_DAI_CLK, + MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_48); + break; + case 64: + regmap_update_bits(max98371->regmap, + MAX98371_DAI_CLK, + MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_64); + break; + default: + pr_info("%s: ratio unsupported %d", __func__, blr_clk_ratio); + return -EINVAL; + } + + switch (rate) { + case 32000: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_32); + break; + case 44100: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_44); + break; + case 48000: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_48); + break; + case 88200: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_88); + break; + case 96000: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_96); + break; + default: + pr_info("%s: rate unsupported %d", __func__, rate); + return -EINVAL; + } + + /* enabling both the RX channels*/ + regmap_update_bits(max98371->regmap, MAX98371_MONOMIX_SRC, + MAX98371_MONOMIX_SRC_MASK, MONOMIX_RX_0_1); + regmap_update_bits(max98371->regmap, MAX98371_DAI_CHANNEL, + MAX98371_CHANNEL_MASK, MAX98371_CHANNEL_MASK); + return 0; +} + +static const struct snd_soc_dapm_widget max98371_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("DAI_OUT", + "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC Enable", + "HiFi Playback", MAX98371_SPK_ENABLE, 0, 0), + SND_SOC_DAPM_SUPPLY("Global Enable", + MAX98371_GLOBAL_ENABLE, 0, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("SPK_OUT"), +}; + +static const struct snd_soc_dapm_route max98371_audio_map[] = { + {"DAC Enable", NULL, "DAI_OUT"}, + {"SPK_OUT", NULL, "DAC Enable"}, + {"SPK_OUT", NULL, "Global Enable"}, +}; + +#define MAX98371_RATES SNDRV_PCM_RATE_8000_48000 +#define MAX98371_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE) + +static struct snd_soc_dai_ops max98371_dai_ops = { + .set_fmt = max98371_dai_set_fmt, + .hw_params = max98371_dai_hw_params, +}; + +static struct snd_soc_dai_driver max98371_dai[] = { + { + .name = "max98371-aif1", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = MAX98371_FORMATS, + }, + .ops = &max98371_dai_ops, + } +}; + +static struct snd_soc_codec_driver max98371_codec = { + .controls = max98371_snd_controls, + .num_controls = ARRAY_SIZE(max98371_snd_controls), + .dapm_routes = max98371_audio_map, + .num_dapm_routes = ARRAY_SIZE(max98371_audio_map), + .dapm_widgets = max98371_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max98371_dapm_widgets), +}; + +static const struct regmap_config max98371_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = MAX98371_VERSION, + .reg_defaults = max98371_reg, + .num_reg_defaults = ARRAY_SIZE(max98371_reg), + .volatile_reg = max98371_volatile_register, + .readable_reg = max98371_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int max98371_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct max98371_priv *max98371; + int ret, reg; + + max98371 = devm_kzalloc(&i2c->dev, + sizeof(*max98371), GFP_KERNEL); + if (!max98371) + return -ENOMEM; + + i2c_set_clientdata(i2c, max98371); + max98371->regmap = devm_regmap_init_i2c(i2c, &max98371_regmap); + if (IS_ERR(max98371->regmap)) { + ret = PTR_ERR(max98371->regmap); + dev_err(&i2c->dev, + "Failed to allocate regmap: %d\n", ret); + return ret; + } + + ret = regmap_read(max98371->regmap, MAX98371_VERSION, &reg); + if (ret < 0) { + dev_info(&i2c->dev, "device error %d\n", ret); + return ret; + } + dev_info(&i2c->dev, "device version %x\n", reg); + + ret = snd_soc_register_codec(&i2c->dev, &max98371_codec, + max98371_dai, ARRAY_SIZE(max98371_dai)); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); + return ret; + } + return ret; +} + +static int max98371_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id max98371_i2c_id[] = { + { "max98371", 0 }, +}; + +MODULE_DEVICE_TABLE(i2c, max98371_i2c_id); + +static const struct of_device_id max98371_of_match[] = { + { .compatible = "maxim,max98371", }, + { } +}; +MODULE_DEVICE_TABLE(of, max98371_of_match); + +static struct i2c_driver max98371_i2c_driver = { + .driver = { + .name = "max98371", + .owner = THIS_MODULE, + .pm = NULL, + .of_match_table = of_match_ptr(max98371_of_match), + }, + .probe = max98371_i2c_probe, + .remove = max98371_i2c_remove, + .id_table = max98371_i2c_id, +}; + +module_i2c_driver(max98371_i2c_driver); + +MODULE_AUTHOR("anish kumar <yesanishhere(a)gmail.com>"); +MODULE_DESCRIPTION("ALSA SoC MAX98371 driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max98371.h b/sound/soc/codecs/max98371.h new file mode 100644 index 0000000..c75b259 --- /dev/null +++ b/sound/soc/codecs/max98371.h @@ -0,0 +1,64 @@ +/* + * max98371.h -- MAX98371 ALSA SoC Audio driver + * + * Copyright 2011-2012 Maxim Integrated Products + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _MAX98371_H +#define _MAX98371_H + +#define MAX98371_IRQ_CLEAR1 0x01 +#define MAX98371_IRQ_CLEAR2 0x02 +#define MAX98371_IRQ_CLEAR3 0x03 +#define MAX98371_DAI_CLK 0x10 +#define MAX98371_DAI_BSEL_MASK 0xF +#define MAX98371_DAI_BSEL_32 2 +#define MAX98371_DAI_BSEL_48 3 +#define MAX98371_DAI_BSEL_64 4 +#define MAX98371_SPK_SR 0x11 +#define MAX98371_SPK_SR_MASK 0xF +#define MAX98371_SPK_SR_32 6 +#define MAX98371_SPK_SR_44 7 +#define MAX98371_SPK_SR_48 8 +#define MAX98371_SPK_SR_88 10 +#define MAX98371_SPK_SR_96 11 +#define MAX98371_DAI_CHANNEL 0x15 +#define MAX98371_CHANNEL_MASK 0x3 +#define MAX98371_MONOMIX_SRC 0x18 +#define MAX98371_MONOMIX_CFG 0x19 +#define MAX98371_HPF 0x1C +#define MAX98371_MONOMIX_SRC_MASK 0xFF +#define MONOMIX_RX_0_1 ((0x1)<<(4)) +#define M98371_DAI_CHANNEL_I2S 0x3 +#define MAX98371_DIGITAL_GAIN 0x2D +#define MAX98371_DIGITAL_GAIN_WIDTH 0x7 +#define MAX98371_GAIN 0x2E +#define MAX98371_GAIN_SHIFT 0x4 +#define MAX98371_GAIN_WIDTH 0x4 +#define MAX98371_NO_LOAD_WIDTH 4 +#define MAX98371_FMT 0x14 +#define MAX98371_CHANSZ_WIDTH 6 +#define MAX98371_FMT_MASK ((0x3)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_FMT_MODE_MASK ((0x7)<<(3)) +#define MAX98371_DAI_LEFT ((0x1)<<(3)) +#define MAX98371_DAI_RIGHT ((0x2)<<(3)) +#define MAX98371_DAI_CHANSZ_16 ((1)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_DAI_CHANSZ_24 ((2)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_DAI_CHANSZ_32 ((3)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_DHT 0x32 +#define MAX98371_DHT_STEP 0x3 +#define MAX98371_SPK_ENABLE 0x4A +#define MAX98371_GLOBAL_ENABLE 0x50 +#define MAX98371_SOFT_RESET 0x51 +#define MAX98371_VERSION 0xFF + + +struct max98371_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; +}; +#endif -- 1.9.3

3 5

[alsa-devel] Real-time audio over local network with good quality
by Ricardo Biehl 10 Apr '16

10 Apr '16

Hello guys! I'm sorry if this mail is not part of the scope of this list :-( I've come from the kernelnewbies mailing list with the same subject as following ... I'm working on a project which aims to be a microphone system that will work over local network <https://github.com/ricardobiehl/ipmic>. So I've found many difficulties in ALSA library and Linux Network. Basically, I need help to make my software get low latency and high quality audio :-) . When I said "high quality audio" I'm referring to (1channel * 16bit sample * 22050Hz) without noise. And "low latency" is about to 0.1 second or less. Any contribution with the project is welcome! Thank you a lot! -- Ricardo Biehl Pasquali

2 2

[alsa-devel] [PATCH 1/2] sound: usb: add UAC2 clock sources as mixer controls
by Daniel Mack 09 Apr '16

09 Apr '16

UAC2 specifies clock sources that optionally have validity controls. This patch exposes them as mixer controls, so they can be read (and at least in theory even be written) by userspace applications in order to make clock selection policy decisions. This implementation does nothing if the device is not UAC2 compliant, or if the clock source does not define said validity control bits. Tested with a miniDSP USBStreamer (0x2752/0x0016), Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- sound/usb/mixer.c | 66 ++++++++++++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 65 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 4f85757..973274b 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -45,6 +45,7 @@ #include <linux/bitops.h> #include <linux/init.h> #include <linux/list.h> +#include <linux/log2.h> #include <linux/slab.h> #include <linux/string.h> #include <linux/usb.h> @@ -1378,6 +1379,68 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, snd_usb_mixer_add_control(&cval->head, kctl); } +static int parse_clock_source_unit(struct mixer_build *state, int unitid, + void *_ftr) + +{ + struct uac_clock_source_descriptor *hdr = _ftr; + struct usb_mixer_elem_info *cval; + struct snd_kcontrol *kctl; + char name[100]; + int ret; + + if (state->mixer->protocol != UAC_VERSION_2) + return -EINVAL; + + if (hdr->bLength != sizeof(*hdr)) + return -EINVAL; + + /* + * The only property of this unit we are interested in is the + * clock source validity. If that isn't readable, just bail out. + */ + if (!uac2_control_is_readable(hdr->bmControls, + ilog2(UAC2_CS_CONTROL_CLOCK_VALID))) + return 0; + + cval = kzalloc(sizeof(*cval), GFP_KERNEL); + if (!cval) + return -ENOMEM; + + snd_usb_mixer_elem_init_std(&cval->head, state->mixer, hdr->bClockID); + + cval->min = 0; + cval->max = 1; + cval->channels = 1; + cval->val_type = USB_MIXER_BOOLEAN; + cval->control = UAC2_CS_CONTROL_CLOCK_VALID; + + if (uac2_control_is_writeable(hdr->bmControls, + ilog2(UAC2_CS_CONTROL_CLOCK_VALID))) + kctl = snd_ctl_new1(&usb_feature_unit_ctl, cval); + else { + cval->master_readonly = 1; + kctl = snd_ctl_new1(&usb_feature_unit_ctl_ro, cval); + } + + if (!kctl) { + kfree(cval); + return -ENOMEM; + } + + kctl->private_free = snd_usb_mixer_elem_free; + ret = snd_usb_copy_string_desc(state, hdr->iClockSource, + name, sizeof(name)); + if (ret > 0) + snprintf(kctl->id.name, sizeof(kctl->id.name), + "%s validity", name); + else + snprintf(kctl->id.name, sizeof(kctl->id.name), + "Clock source %d validity", hdr->bClockID); + + return snd_usb_mixer_add_control(&cval->head, kctl); +} + /* * parse a feature unit * @@ -2126,10 +2189,11 @@ static int parse_audio_unit(struct mixer_build *state, int unitid) switch (p1[2]) { case UAC_INPUT_TERMINAL: - case UAC2_CLOCK_SOURCE: return 0; /* NOP */ case UAC_MIXER_UNIT: return parse_audio_mixer_unit(state, unitid, p1); + case UAC2_CLOCK_SOURCE: + return parse_clock_source_unit(state, unitid, p1); case UAC_SELECTOR_UNIT: case UAC2_CLOCK_SELECTOR: return parse_audio_selector_unit(state, unitid, p1); -- 2.4.3

2 8

[alsa-devel] [PATCH v2 1/2] sound: usb: add UAC2 clock sources as mixer controls
by Daniel Mack 09 Apr '16

09 Apr '16

UAC2 specifies clock sources that optionally have validity controls. This patch exposes them as mixer controls, so they can be read (and at least in theory even be written) by userspace applications in order to make clock selection policy decisions. This implementation does nothing if the device is not UAC2 compliant, or if the clock source does not define said validity control bits. Tested with a miniDSP USBStreamer (0x2752/0x0016). Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- sound/usb/mixer.c | 69 ++++++++++++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 68 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 4f85757..cab6f52 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -45,6 +45,7 @@ #include <linux/bitops.h> #include <linux/init.h> #include <linux/list.h> +#include <linux/log2.h> #include <linux/slab.h> #include <linux/string.h> #include <linux/usb.h> @@ -1378,6 +1379,71 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, snd_usb_mixer_add_control(&cval->head, kctl); } +static int parse_clock_source_unit(struct mixer_build *state, int unitid, + void *_ftr) +{ + struct uac_clock_source_descriptor *hdr = _ftr; + struct usb_mixer_elem_info *cval; + struct snd_kcontrol *kctl; + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + int ret; + + if (state->mixer->protocol != UAC_VERSION_2) + return -EINVAL; + + if (hdr->bLength != sizeof(*hdr)) { + usb_audio_dbg(state->chip, + "Bogus clock source descriptor length of %d, ignoring.\n", + hdr->bLength); + return 0; + } + + /* + * The only property of this unit we are interested in is the + * clock source validity. If that isn't readable, just bail out. + */ + if (!uac2_control_is_readable(hdr->bmControls, + ilog2(UAC2_CS_CONTROL_CLOCK_VALID))) + return 0; + + cval = kzalloc(sizeof(*cval), GFP_KERNEL); + if (!cval) + return -ENOMEM; + + snd_usb_mixer_elem_init_std(&cval->head, state->mixer, hdr->bClockID); + + cval->min = 0; + cval->max = 1; + cval->channels = 1; + cval->val_type = USB_MIXER_BOOLEAN; + cval->control = UAC2_CS_CONTROL_CLOCK_VALID; + + if (uac2_control_is_writeable(hdr->bmControls, + ilog2(UAC2_CS_CONTROL_CLOCK_VALID))) + kctl = snd_ctl_new1(&usb_feature_unit_ctl, cval); + else { + cval->master_readonly = 1; + kctl = snd_ctl_new1(&usb_feature_unit_ctl_ro, cval); + } + + if (!kctl) { + kfree(cval); + return -ENOMEM; + } + + kctl->private_free = snd_usb_mixer_elem_free; + ret = snd_usb_copy_string_desc(state, hdr->iClockSource, + name, sizeof(name)); + if (ret > 0) + snprintf(kctl->id.name, sizeof(kctl->id.name), + "%s Validity", name); + else + snprintf(kctl->id.name, sizeof(kctl->id.name), + "Clock Source %d Validity", hdr->bClockID); + + return snd_usb_mixer_add_control(&cval->head, kctl); +} + /* * parse a feature unit * @@ -2126,10 +2192,11 @@ static int parse_audio_unit(struct mixer_build *state, int unitid) switch (p1[2]) { case UAC_INPUT_TERMINAL: - case UAC2_CLOCK_SOURCE: return 0; /* NOP */ case UAC_MIXER_UNIT: return parse_audio_mixer_unit(state, unitid, p1); + case UAC2_CLOCK_SOURCE: + return parse_clock_source_unit(state, unitid, p1); case UAC_SELECTOR_UNIT: case UAC2_CLOCK_SELECTOR: return parse_audio_selector_unit(state, unitid, p1); -- 2.4.3

1 1

[alsa-devel] [PATCH] ALSA: constify ct_timer_ops structures
by Julia Lawall 09 Apr '16

09 Apr '16

The ct_timer_ops structures are never modified, so declare them as const. Done with the help of Coccinelle. Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- sound/pci/ctxfi/cttimer.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/ctxfi/cttimer.c b/sound/pci/ctxfi/cttimer.c index a5d4604..8f94534 100644 --- a/sound/pci/ctxfi/cttimer.c +++ b/sound/pci/ctxfi/cttimer.c @@ -49,7 +49,7 @@ struct ct_timer { spinlock_t lock; /* global timer lock (for xfitimer) */ spinlock_t list_lock; /* lock for instance list */ struct ct_atc *atc; - struct ct_timer_ops *ops; + const struct ct_timer_ops *ops; struct list_head instance_head; struct list_head running_head; unsigned int wc; /* current wallclock */ @@ -128,7 +128,7 @@ static void ct_systimer_prepare(struct ct_timer_instance *ti) #define ct_systimer_free ct_systimer_prepare -static struct ct_timer_ops ct_systimer_ops = { +static const struct ct_timer_ops ct_systimer_ops = { .init = ct_systimer_init, .free_instance = ct_systimer_free, .prepare = ct_systimer_prepare, @@ -322,7 +322,7 @@ static void ct_xfitimer_free_global(struct ct_timer *atimer) ct_xfitimer_irq_stop(atimer); } -static struct ct_timer_ops ct_xfitimer_ops = { +static const struct ct_timer_ops ct_xfitimer_ops = { .prepare = ct_xfitimer_prepare, .start = ct_xfitimer_start, .stop = ct_xfitimer_stop,

2 1

[alsa-devel] [PATCH] ASoC: samsung: Remove unused Odroid x2/u3 machine driver
by Sylwester Nawrocki 08 Apr '16

08 Apr '16

We have been using "simple-audio-card" for Odroid X2/U3 boards, as can be seen from sound node in arch/arm/boot/dts/ exynos4412-odroid-common.dtsi. A dedicated machine driver is not needed and it is removed in this patch. There is no dts files using "samsung,odroidx2-audio" or "samsung,odroidu3-audio" compatible strings. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- .../bindings/sound/samsung,odroidx2-max98090.txt | 35 ---- sound/soc/samsung/Kconfig | 8 - sound/soc/samsung/Makefile | 2 - sound/soc/samsung/odroidx2_max98090.c | 185 --------------------- 4 files changed, 230 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/samsung,odroidx2-max98090.txt delete mode 100644 sound/soc/samsung/odroidx2_max98090.c diff --git a/Documentation/devicetree/bindings/sound/samsung,odroidx2-max98090.txt b/Documentation/devicetree/bindings/sound/samsung,odroidx2-max98090.txt deleted file mode 100644 index 9148f72..0000000 --- a/Documentation/devicetree/bindings/sound/samsung,odroidx2-max98090.txt +++ /dev/null @@ -1,35 +0,0 @@ -Samsung Exynos Odroid X2/U3 audio complex with MAX98090 codec - -Required properties: - - compatible : "samsung,odroidx2-audio" - for Odroid X2 board, - "samsung,odroidu3-audio" - for Odroid U3 board - - samsung,model : the user-visible name of this sound complex - - samsung,i2s-controller : the phandle of the I2S controller - - samsung,audio-codec : the phandle of the MAX98090 audio codec - - samsung,audio-routing : a list of the connections between audio - components; each entry is a pair of strings, the first being the - connection's sink, the second being the connection's source; - valid names for sources and sinks are the MAX98090's pins (as - documented in its binding), and the jacks on the board - For Odroid X2: - * Headphone Jack - * Mic Jack - * DMIC - - For Odroid U3: - * Headphone Jack - * Speakers - -Example: - -sound { - compatible = "samsung,odroidu3-audio"; - samsung,i2s-controller = <&i2s0>; - samsung,audio-codec = <&max98090>; - samsung,model = "Odroid-X2"; - samsung,audio-routing = - "Headphone Jack", "HPL", - "Headphone Jack", "HPR", - "IN1", "Mic Jack", - "Mic Jack", "MICBIAS"; -}; diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 78baa26..7b722b0 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -224,14 +224,6 @@ config SND_SOC_SNOW Say Y if you want to add audio support for various Snow boards based on Exynos5 series of SoCs. -config SND_SOC_ODROIDX2 - tristate "Audio support for Odroid-X2 and Odroid-U3" - depends on SND_SOC_SAMSUNG && I2C - select SND_SOC_MAX98090 - select SND_SAMSUNG_I2S - help - Say Y here to enable audio support for the Odroid-X2/U3. - config SND_SOC_ARNDALE_RT5631_ALC5631 tristate "Audio support for RT5631(ALC5631) on Arndale Board" depends on SND_SOC_SAMSUNG && I2C diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 052fe71..5d03f5c 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -43,7 +43,6 @@ snd-soc-tobermory-objs := tobermory.o snd-soc-lowland-objs := lowland.o snd-soc-littlemill-objs := littlemill.o snd-soc-bells-objs := bells.o -snd-soc-odroidx2-max98090-objs := odroidx2_max98090.o snd-soc-arndale-rt5631-objs := arndale_rt5631.o obj-$(CONFIG_SND_SOC_SAMSUNG_JIVE_WM8750) += snd-soc-jive-wm8750.o @@ -69,5 +68,4 @@ obj-$(CONFIG_SND_SOC_TOBERMORY) += snd-soc-tobermory.o obj-$(CONFIG_SND_SOC_LOWLAND) += snd-soc-lowland.o obj-$(CONFIG_SND_SOC_LITTLEMILL) += snd-soc-littlemill.o obj-$(CONFIG_SND_SOC_BELLS) += snd-soc-bells.o -obj-$(CONFIG_SND_SOC_ODROIDX2) += snd-soc-odroidx2-max98090.o obj-$(CONFIG_SND_SOC_ARNDALE_RT5631_ALC5631) += snd-soc-arndale-rt5631.o diff --git a/sound/soc/samsung/odroidx2_max98090.c b/sound/soc/samsung/odroidx2_max98090.c deleted file mode 100644 index 0421727..0000000 --- a/sound/soc/samsung/odroidx2_max98090.c +++ /dev/null @@ -1,185 +0,0 @@ -/* - * Copyright (C) 2014 Samsung Electronics Co., Ltd. - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ - -#include <linux/of.h> -#include <linux/module.h> -#include <sound/soc.h> -#include <sound/pcm_params.h> -#include "i2s.h" - -struct odroidx2_drv_data { - const struct snd_soc_dapm_widget *dapm_widgets; - unsigned int num_dapm_widgets; -}; - -/* The I2S CDCLK output clock frequency for the MAX98090 codec */ -#define MAX98090_MCLK 19200000 - -static struct snd_soc_dai_link odroidx2_dai[]; - -static int odroidx2_late_probe(struct snd_soc_card *card) -{ - struct snd_soc_pcm_runtime *rtd; - struct snd_soc_dai *codec_dai; - struct snd_soc_dai *cpu_dai; - int ret; - - rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); - codec_dai = rtd->codec_dai; - cpu_dai = rtd->cpu_dai; - - ret = snd_soc_dai_set_sysclk(codec_dai, 0, MAX98090_MCLK, - SND_SOC_CLOCK_IN); - - if (ret < 0 || of_find_property(odroidx2_dai[0].codec_of_node, - "clocks", NULL)) - return ret; - - /* Set the cpu DAI configuration in order to use CDCLK */ - return snd_soc_dai_set_sysclk(cpu_dai, SAMSUNG_I2S_CDCLK, - 0, SND_SOC_CLOCK_OUT); -} - -static const struct snd_soc_dapm_widget odroidx2_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_MIC("Mic Jack", NULL), - SND_SOC_DAPM_MIC("DMIC", NULL), -}; - -static const struct snd_soc_dapm_widget odroidu3_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_SPK("Speakers", NULL), -}; - -static struct snd_soc_dai_link odroidx2_dai[] = { - { - .name = "MAX98090", - .stream_name = "MAX98090 PCM", - .codec_dai_name = "HiFi", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, - } -}; - -static struct snd_soc_card odroidx2 = { - .owner = THIS_MODULE, - .dai_link = odroidx2_dai, - .num_links = ARRAY_SIZE(odroidx2_dai), - .fully_routed = true, - .late_probe = odroidx2_late_probe, -}; - -static const struct odroidx2_drv_data odroidx2_drvdata = { - .dapm_widgets = odroidx2_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(odroidx2_dapm_widgets), -}; - -static const struct odroidx2_drv_data odroidu3_drvdata = { - .dapm_widgets = odroidu3_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(odroidu3_dapm_widgets), -}; - -static const struct of_device_id odroidx2_audio_of_match[] = { - { - .compatible = "samsung,odroidx2-audio", - .data = &odroidx2_drvdata, - }, { - .compatible = "samsung,odroidu3-audio", - .data = &odroidu3_drvdata, - }, - { }, -}; -MODULE_DEVICE_TABLE(of, odroidx2_audio_of_match); - -static int odroidx2_audio_probe(struct platform_device *pdev) -{ - struct device_node *snd_node = pdev->dev.of_node; - struct snd_soc_card *card = &odroidx2; - struct device_node *i2s_node, *codec_node; - struct odroidx2_drv_data *dd; - const struct of_device_id *of_id; - int ret; - - of_id = of_match_node(odroidx2_audio_of_match, snd_node); - dd = (struct odroidx2_drv_data *)of_id->data; - - card->num_dapm_widgets = dd->num_dapm_widgets; - card->dapm_widgets = dd->dapm_widgets; - - card->dev = &pdev->dev; - - ret = snd_soc_of_parse_card_name(card, "samsung,model"); - if (ret < 0) - return ret; - - ret = snd_soc_of_parse_audio_routing(card, "samsung,audio-routing"); - if (ret < 0) - return ret; - - codec_node = of_parse_phandle(snd_node, "samsung,audio-codec", 0); - if (!codec_node) { - dev_err(&pdev->dev, - "Failed parsing samsung,i2s-codec property\n"); - return -EINVAL; - } - - i2s_node = of_parse_phandle(snd_node, "samsung,i2s-controller", 0); - if (!i2s_node) { - dev_err(&pdev->dev, - "Failed parsing samsung,i2s-controller property\n"); - ret = -EINVAL; - goto err_put_codec_n; - } - - odroidx2_dai[0].codec_of_node = codec_node; - odroidx2_dai[0].cpu_of_node = i2s_node; - odroidx2_dai[0].platform_of_node = i2s_node; - - ret = snd_soc_register_card(card); - if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", - ret); - goto err_put_i2s_n; - } - return 0; - -err_put_i2s_n: - of_node_put(i2s_node); -err_put_codec_n: - of_node_put(codec_node); - return ret; -} - -static int odroidx2_audio_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - - of_node_put(odroidx2_dai[0].cpu_of_node); - of_node_put(odroidx2_dai[0].codec_of_node); - - return 0; -} - -static struct platform_driver odroidx2_audio_driver = { - .driver = { - .name = "odroidx2-audio", - .of_match_table = odroidx2_audio_of_match, - .pm = &snd_soc_pm_ops, - }, - .probe = odroidx2_audio_probe, - .remove = odroidx2_audio_remove, -}; -module_platform_driver(odroidx2_audio_driver); - -MODULE_AUTHOR("Chen Zhen <zhen1.chen(a)samsung.com>"); -MODULE_AUTHOR("Sylwester Nawrocki <s.nawrocki(a)samsung.com>"); -MODULE_DESCRIPTION("ALSA SoC Odroid X2/U3 Audio Support"); -MODULE_LICENSE("GPL v2"); -- 1.9.1

1 0

[alsa-devel] [PATCH 1/3] ASoC: samsung: Remove unused "samsung-i2sv4" platform_device_id entry
by Sylwester Nawrocki 08 Apr '16

08 Apr '16

"samsung-i2sv4" identifier was previously used for the I2S device of the S5PV210 SoCs, it can be removed now when s5pv210 is a dt-only platform. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/samsung/i2s.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 70a2559..ededac9 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1492,9 +1492,6 @@ static const struct platform_device_id samsung_i2s_driver_ids[] = { }, { .name = "samsung-i2s-sec", .driver_data = (kernel_ulong_t)&samsung_dai_type_sec, - }, { - .name = "samsung-i2sv4", - .driver_data = (kernel_ulong_t)&i2sv5_dai_type, }, {}, }; -- 1.9.1

1 2

[alsa-devel] [PATCH] ASoC: arizona: Do not create OUT4R widget for CS47L24/WM1831
by Richard Fitzgerald 08 Apr '16

08 Apr '16

The CS47L24 and WM1831 codecs only use the OUT4L widget so we can skip creation of the OUT4R widget. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 92d22a0..d8a6823 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -221,6 +221,8 @@ int arizona_init_spk(struct snd_soc_codec *codec) switch (arizona->type) { case WM8997: + case CS47L24: + case WM1831: break; default: ret = snd_soc_dapm_new_controls(dapm, &arizona_spkr, 1); -- 1.9.1

1 0

[alsa-devel] [PATCH v2 0/6] Propagate errors out from compressed streams
by Charles Keepax 08 Apr '16

08 Apr '16

If the DSP suffers an unrecoverable error, the driver likely knows about this, however the framework may not get informed because errors returned from pointer requests are ignored within the framework. Things work out fine if user-space is doing a read as reads return error status back to user-space so the user can find out that things have gone bad. However, if user-space is doing an avail request there is no path for the error to come back up to user-space. The pointer request returns zero available data, so a read never happens and we basically just end up sitting waiting for data on a stream that we know full well has died. This patch set attempts to address this and ensure that errors are fully propagated to user-space and we don't ever end up wait for data that will never come. Changes since v1: - Based on a recommendation from Vinod, rather than handling errors from pointer requests, add a new call that allows us to inform the compress core that the stream has gone bad. Then return errors from the avail and poll callbacks if the stream is bad. Thanks, Charles Charles Keepax (6): ASoC: compress: Pass error out of soc_compr_pointer ALSA: compress: Add function to indicate the stream has gone bad ALSA: compress: Replace complex if statement with switch ASoC: wm_adsp: Factor out fetching of stream errors from the DSP ASoC: wm_adsp: Improve DSP error handling ASoC: wm_adsp: Use new snd_compr_stop_xrun to signal stream failure include/sound/compress_driver.h | 3 ++ sound/core/compress_offload.c | 67 ++++++++++++++++++++++++++++++++++++++--- sound/soc/codecs/wm_adsp.c | 43 +++++++++++++++++++------- sound/soc/soc-compress.c | 5 +-- 4 files changed, 101 insertions(+), 17 deletions(-) -- 2.1.4

4 13

[alsa-devel] [PATCH] [linux-next]:ALSA: Fix a typo in timestamping.txt
by Masanari Iida 08 Apr '16

08 Apr '16

This patch fix a spelling typo found in Documentation/sound/alsa/timestamping.txt Signed-off-by: Masanari Iida <standby24x7(a)gmail.com> --- Documentation/sound/alsa/timestamping.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/sound/alsa/timestamping.txt b/Documentation/sound/alsa/timestamping.txt index 0b191a23f534..1b6473f393a8 100644 --- a/Documentation/sound/alsa/timestamping.txt +++ b/Documentation/sound/alsa/timestamping.txt @@ -129,7 +129,7 @@ will be required to issue multiple queries and perform an interpolation of the results In some hardware-specific configuration, the system timestamp is -latched by a low-level audio subsytem, and the information provided +latched by a low-level audio subsystem, and the information provided back to the driver. Due to potential delays in the communication with the hardware, there is a risk of misalignment with the avail and delay information. To make sure applications are not confused, a -- 2.8.0

2 1

[alsa-devel] [PATCH v3 0/7] topology: Add support for vendor tuples
by mengdong.lin＠linux.intel.com 07 Apr '16

07 Apr '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This series addes support for vendor tuples to topology, to avoid importing binary data blob from other files. Backward compatibility of ABI is not impacted. A kernel patch is also submitted "ASoC: topology: ABI - Define types for vendor tuples". The 1st patch is small code cleanup. The 2nd patch is a preparation, since tuples will need the type-specific free handler. History: v2: add check on string length, use strtol() to get hex value, and fix memory leak. v3: Fix overflow on uuid string copy. Use strtoul() to get unsigned vendor values and check range, tested on 32 & 64 bit systems. Mengdong Lin (7): topology: Use the generic pointer to free an element's object topology: Define a free handler for the element topology: Add doc for vendor tuples topology: ABI - Define types for vendor tuples topology: Add support for vendor tokens topology: Add support for parsing vendor tuples topology: Build data objects with tuples include/sound/asoc.h | 42 +++- include/topology.h | 79 +++++++- src/topology/data.c | 503 +++++++++++++++++++++++++++++++++++++++++++++- src/topology/elem.c | 15 +- src/topology/parser.c | 24 +++ src/topology/tplg_local.h | 47 +++++ 6 files changed, 702 insertions(+), 8 deletions(-) -- 2.5.0

3 10

[alsa-devel] [PATCH] pcm_plugin: fix appl pointer not correct when mmap_commit() return error
by Shengjiu Wang 07 Apr '16

07 Apr '16

When snd_pcm_mmap_commit() return error, the appl pointer is also updated. which cause the avail_update()'s result wrong. This patch move the snd_pcm_mmap_appl_forward() to the place when snd_pcm_mmap_commit() is successfully returned. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)freescale.com> --- src/pcm/pcm_plugin.c | 48 ++++++++++++++++++++++++++++++++---------------- 1 file changed, 32 insertions(+), 16 deletions(-) diff --git a/src/pcm/pcm_plugin.c b/src/pcm/pcm_plugin.c index d007e8c..940491d 100644 --- a/src/pcm/pcm_plugin.c +++ b/src/pcm/pcm_plugin.c @@ -279,18 +279,22 @@ static snd_pcm_sframes_t snd_pcm_plugin_write_areas(snd_pcm_t *pcm, return -EPIPE; } snd_atomic_write_begin(&plugin->watom); - snd_pcm_mmap_appl_forward(pcm, frames); result = snd_pcm_mmap_commit(slave, slave_offset, slave_frames); if (result > 0 && (snd_pcm_uframes_t)result != slave_frames) { snd_pcm_sframes_t res; res = plugin->undo_write(pcm, slave_areas, slave_offset + result, slave_frames, slave_frames - result); - if (res < 0) + if (res < 0) { + snd_atomic_write_end(&plugin->watom); return xfer > 0 ? (snd_pcm_sframes_t)xfer : res; + } frames -= res; } - snd_atomic_write_end(&plugin->watom); - if (result <= 0) + if (result <= 0) { + snd_atomic_write_end(&plugin->watom); return xfer > 0 ? (snd_pcm_sframes_t)xfer : result; + } + snd_pcm_mmap_appl_forward(pcm, frames); + snd_atomic_write_end(&plugin->watom); offset += frames; xfer += frames; size -= frames; @@ -325,19 +329,23 @@ static snd_pcm_sframes_t snd_pcm_plugin_read_areas(snd_pcm_t *pcm, return -EPIPE; } snd_atomic_write_begin(&plugin->watom); - snd_pcm_mmap_appl_forward(pcm, frames); result = snd_pcm_mmap_commit(slave, slave_offset, slave_frames); if (result > 0 && (snd_pcm_uframes_t)result != slave_frames) { snd_pcm_sframes_t res; res = plugin->undo_read(slave, areas, offset, frames, slave_frames - result); - if (res < 0) + if (res < 0) { + snd_atomic_write_end(&plugin->watom); return xfer > 0 ? (snd_pcm_sframes_t)xfer : res; + } frames -= res; } - snd_atomic_write_end(&plugin->watom); - if (result <= 0) + if (result <= 0) { + snd_atomic_write_end(&plugin->watom); return xfer > 0 ? (snd_pcm_sframes_t)xfer : result; + } + snd_pcm_mmap_appl_forward(pcm, frames); + snd_atomic_write_end(&plugin->watom); offset += frames; xfer += frames; size -= frames; @@ -423,19 +431,23 @@ snd_pcm_plugin_mmap_commit(snd_pcm_t *pcm, frames = plugin->write(pcm, areas, appl_offset, frames, slave_areas, slave_offset, &slave_frames); snd_atomic_write_begin(&plugin->watom); - snd_pcm_mmap_appl_forward(pcm, frames); result = snd_pcm_mmap_commit(slave, slave_offset, slave_frames); - snd_atomic_write_end(&plugin->watom); if (result > 0 && (snd_pcm_uframes_t)result != slave_frames) { snd_pcm_sframes_t res; res = plugin->undo_write(pcm, slave_areas, slave_offset + result, slave_frames, slave_frames - result); - if (res < 0) + if (res < 0) { + snd_atomic_write_end(&plugin->watom); return xfer > 0 ? xfer : res; + } frames -= res; } - if (result <= 0) + if (result <= 0) { + snd_atomic_write_end(&plugin->watom); return xfer > 0 ? xfer : result; + } + snd_pcm_mmap_appl_forward(pcm, frames); + snd_atomic_write_end(&plugin->watom); if (frames == cont) appl_offset = 0; else @@ -490,19 +502,23 @@ static snd_pcm_sframes_t snd_pcm_plugin_avail_update(snd_pcm_t *pcm) frames = (plugin->read)(pcm, areas, hw_offset, frames, slave_areas, slave_offset, &slave_frames); snd_atomic_write_begin(&plugin->watom); - snd_pcm_mmap_hw_forward(pcm, frames); result = snd_pcm_mmap_commit(slave, slave_offset, slave_frames); - snd_atomic_write_end(&plugin->watom); if (result > 0 && (snd_pcm_uframes_t)result != slave_frames) { snd_pcm_sframes_t res; res = plugin->undo_read(slave, areas, hw_offset, frames, slave_frames - result); - if (res < 0) + if (res < 0) { + snd_atomic_write_end(&plugin->watom); return xfer > 0 ? (snd_pcm_sframes_t)xfer : res; + } frames -= res; } - if (result <= 0) + if (result <= 0) { + snd_atomic_write_end(&plugin->watom); return xfer > 0 ? (snd_pcm_sframes_t)xfer : result; + } + snd_pcm_mmap_hw_forward(pcm, frames); + snd_atomic_write_end(&plugin->watom); if (frames == cont) hw_offset = 0; else -- 1.9.1

2 1

[alsa-devel] Synchonize video with audio playback
by Li, Steve 07 Apr '16

07 Apr '16

Hi, I am new to ALSA. I am trying to implement video and audio playback in my app. I am able to find a lot of sample code for audio playback, but I am wondering what is the best way to synchronize video playback with audio. What kind of timing information can I query through the ALSA interface in terms of the audio playback? Your help is appreciated. Thanks Steve ______________________________________________________________________ This email has been scanned by the Symantec Email Security.cloud service. For more information please visit http://www.symanteccloud.com ______________________________________________________________________

2 1

[alsa-devel] [PATCH V7 0/2] ASoC: Add core API to register and cleanup DMI names for card
by han.lu＠intel.com 07 Apr '16

07 Apr '16

From: "Lu, Han" <han.lu(a)intel.com> Share more product information, for user space utils such as PA and UCM to distinguish different products. 1. Add core APIs to register and cleanup DMI names for card. 2. Apply the APIs to bytcr-rt5640 driver. changes on V7: 1. Remove inconsistent API description changes on V6: 1. Use dynamic allocate and cleanup for card long name 2. Remove unneccessary arguments to simplify the API changes on V5: 1. Use independent space to store card long_name, to avoid irrelavant info sharing from card component 2. Use letter ';' instead of ':' to separate strings in long name, in case name strings may also contain ':' and confuse user 3. Fix error that vendor name and firmware name were not optional changes on V4: 1. Replace kmalloc() and snprintf() with ksaprintf() to simplify code changes on V3: 1. Split the core API and the API call to two patches 2. Replace misused strcat() with snprintf() 3. Code and comment fix Lu, Han (2): ASoC: core: add API for registering and cleaning up DMI card names ASoC: bytcr-rt5640: register DMI names for card include/sound/soc.h | 3 ++ sound/soc/intel/boards/bytcr_rt5640.c | 18 ++++++++ sound/soc/soc-core.c | 85 +++++++++++++++++++++++++++++++++++ 3 files changed, 106 insertions(+) -- 2.5.0

5 8

[alsa-devel] [patch] ASoC: Intel: sst: fix a loop timeout in sst_hsw_stream_reset()
by Dan Carpenter 07 Apr '16

07 Apr '16

In the original code we ended the loop with tries set to -1 instead of zero. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- Not tested. diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c index ac60f13..9156522 100644 --- a/sound/soc/intel/haswell/sst-haswell-ipc.c +++ b/sound/soc/intel/haswell/sst-haswell-ipc.c @@ -1345,7 +1345,7 @@ int sst_hsw_stream_reset(struct sst_hsw *hsw, struct sst_hsw_stream *stream) return 0; /* wait for pause to complete before we reset the stream */ - while (stream->running && tries--) + while (stream->running && --tries) msleep(1); if (!tries) { dev_err(hsw->dev, "error: reset stream %d still running\n",

3 2

[alsa-devel] CircuitCo AudioCape and 4.4.6-bone-rt-r7 kernel, not quite there
by Rick Mann 07 Apr '16

07 Apr '16

I'm thinking my problems are more ALSA-related than Beaglebone-specific. I wonder if anyone can help. With kernel 4.4.6-bone-rt-r7, and the CircuitCo AudioCape, I get a sound card in the system, but speaker-test reports errors (Unable to set hw params for playback: Invalid argument). I may have issues in various ALSA config settings; I'm hoping someone here can help me by showing me what to look at and change. The CODEC is a tlv320aic3104. Thanks! ----------- # aplay -l **** List of PLAYBACK Hardware Devices **** card 0: B [AudioCape Rev B], device 0: davinci-mcasp.0-tlv320aic3x-hifi tlv320aic3x-hifi-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 ----------- # speaker-test speaker-test 1.0.28 Playback device is default Stream parameters are 48000Hz, S16_LE, 1 channels Using 16 octaves of pink noise Rate set to 48000Hz (requested 48000Hz) Buffer size range from 16 to 32768 Period size range from 8 to 16384 Using max buffer size 32768 Periods = 4 Unable to set hw params for playback: Invalid argument Setting of hwparams failed: Invalid argument ----- More info: My DTB: http://pastebin.com/Wja7ZQRk and DTBO: https://github.com/RobertCNelson/bb.org-overlays/blob/master/src/arm/BB-BON… Results of speaker-test, aplay, etc.: http://pastebin.com/mMPbReAV One possibility, ALSA is still not configured correctly for this cape, but I wonder if anyone has the right configuration for it? Thanks! -- Rick Mann rmann(a)latencyzero.com

1 0

[alsa-devel] [PATCH] ASoC: mediatek: Add second I2S on mt8173-rt5650 machine driver
by PC Liao 06 Apr '16

06 Apr '16

This patch adds second I2S for rt5650 codec on mt8173-rt5650 machine driver. Signed-off-by: PC Liao <pc.liao(a)mediatek.com> --- .../devicetree/bindings/sound/mt8173-rt5650.txt | 6 +++ sound/soc/mediatek/mt8173-rt5650.c | 55 ++++++++++++++++++-- 2 files changed, 56 insertions(+), 5 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt index fe5a5ef..0046926 100644 --- a/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt +++ b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt @@ -5,11 +5,17 @@ Required properties: - mediatek,audio-codec: the phandles of rt5650 codecs - mediatek,platform: the phandle of MT8173 ASoC platform +Optional properties: +- mediatek,rt5650_i2s: I2S mode of rt5650 + 0: using I2S1 on rt5650 for record + 1: using I2S2 on rt5650 for record + Example: sound { compatible = "mediatek,mt8173-rt5650"; mediatek,audio-codec = <&rt5650>; mediatek,platform = <&afe>; + mediatek,rt5650_i2s = <0>; }; diff --git a/sound/soc/mediatek/mt8173-rt5650.c b/sound/soc/mediatek/mt8173-rt5650.c index bb09bb1..ab4c806 100644 --- a/sound/soc/mediatek/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173-rt5650.c @@ -23,6 +23,20 @@ #define MCLK_FOR_CODECS 12288000 +struct mt8173_rt5650_platform_data { + unsigned int rt5650_i2s; + /* 0 = I2S1; 1 = I2S2 */ +}; + +static struct mt8173_rt5650_platform_data mt8173_rt5650_priv = { + .rt5650_i2s = 0, +}; + +enum mt8173_rt5650_i2s { + MT8173_RT5650_I2S1 = 0, + MT8173_RT5650_I2S2, +}; + static const struct snd_soc_dapm_widget mt8173_rt5650_widgets[] = { SND_SOC_DAPM_SPK("Speaker", NULL), SND_SOC_DAPM_MIC("Int Mic", NULL), @@ -87,10 +101,25 @@ static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime) struct snd_soc_codec *codec = runtime->codec_dais[0]->codec; int ret; - rt5645_sel_asrc_clk_src(codec, - RT5645_DA_STEREO_FILTER | - RT5645_AD_STEREO_FILTER, - RT5645_CLK_SEL_I2S1_ASRC); + switch (mt8173_rt5650_priv.rt5650_i2s) { + case MT8173_RT5650_I2S1: + rt5645_sel_asrc_clk_src(codec, + RT5645_DA_STEREO_FILTER | + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + break; + case MT8173_RT5650_I2S2: + rt5645_sel_asrc_clk_src(codec, + RT5645_DA_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + rt5645_sel_asrc_clk_src(codec, + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S2_ASRC); + break; + default: + break; + } + /* enable jack detection */ ret = snd_soc_card_jack_new(card, "Headset Jack", SND_JACK_HEADPHONE | SND_JACK_MICROPHONE | @@ -112,6 +141,9 @@ static struct snd_soc_dai_link_component mt8173_rt5650_codecs[] = { { .dai_name = "rt5645-aif1", }, + { + .dai_name = "rt5645-aif2", + }, }; enum { @@ -149,7 +181,7 @@ static struct snd_soc_dai_link mt8173_rt5650_dais[] = { .cpu_dai_name = "I2S", .no_pcm = 1, .codecs = mt8173_rt5650_codecs, - .num_codecs = 1, + .num_codecs = 2, .init = mt8173_rt5650_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, @@ -199,6 +231,19 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) "Property 'audio-codec' missing or invalid\n"); return -EINVAL; } + mt8173_rt5650_codecs[1].of_node = mt8173_rt5650_codecs[0].of_node; + + if (device_property_present(&pdev->dev, "mediatek,rt5650_i2s")) { + ret = device_property_read_u32(&pdev->dev, + "mediatek,rt5650_i2s", + &mt8173_rt5650_priv.rt5650_i2s); + if (ret) { + dev_err(&pdev->dev, + "%s snd_soc_register_card fail %d\n", + __func__, ret); + } + } + card->dev = &pdev->dev; platform_set_drvdata(pdev, card); -- 1.7.9.5

2 7

[alsa-devel] [PATCH v2] ASoC: rt5514: add rt5514 SPI driver
by Oder Chiou 06 Apr '16

06 Apr '16

The device has multiple control interfaces, I2C and SPI. The I2C interface mainly controls the register settings of codec. The SPI interface is in order to provide the high speed transmission of data. For example, high bandwidth memory read/write of DSP. The patch adds the rt5514 SPI driver for loading the firmware of DSP and retrieving the voice data from DSP after the system is waked up by specific voice. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt5514-spi.c | 458 ++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5514-spi.h | 38 ++++ sound/soc/codecs/rt5514.c | 249 +++++++++++++++++++++++ sound/soc/codecs/rt5514.h | 4 + 6 files changed, 754 insertions(+) create mode 100644 sound/soc/codecs/rt5514-spi.c create mode 100644 sound/soc/codecs/rt5514-spi.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2f99b11..306c4e3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -627,6 +627,9 @@ config SND_SOC_RT298 config SND_SOC_RT5514 tristate +config SND_SOC_RT5514_SPI + tristate + config SND_SOC_RT5616 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 185a712..501bf43 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -97,6 +97,7 @@ snd-soc-rl6347a-objs := rl6347a.o snd-soc-rt286-objs := rt286.o snd-soc-rt298-objs := rt298.o snd-soc-rt5514-objs := rt5514.o +snd-soc-rt5514-spi-objs := rt5514-spi.o snd-soc-rt5616-objs := rt5616.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o @@ -306,6 +307,7 @@ obj-$(CONFIG_SND_SOC_RL6347A) += snd-soc-rl6347a.o obj-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o obj-$(CONFIG_SND_SOC_RT298) += snd-soc-rt298.o obj-$(CONFIG_SND_SOC_RT5514) += snd-soc-rt5514.o +obj-$(CONFIG_SND_SOC_RT5514_SPI) += snd-soc-rt5514-spi.o obj-$(CONFIG_SND_SOC_RT5616) += snd-soc-rt5616.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c new file mode 100644 index 0000000..631865d --- /dev/null +++ b/sound/soc/codecs/rt5514-spi.c @@ -0,0 +1,458 @@ +/* + * rt5514-spi.c -- RT5514 SPI driver + * + * Copyright 2015 Realtek Semiconductor Corp. + * Author: Oder Chiou <oder_chiou(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/input.h> +#include <linux/spi/spi.h> +#include <linux/device.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/interrupt.h> +#include <linux/irq.h> +#include <linux/slab.h> +#include <linux/gpio.h> +#include <linux/sched.h> +#include <linux/kthread.h> +#include <linux/uaccess.h> +#include <linux/miscdevice.h> +#include <linux/regulator/consumer.h> +#include <linux/pm_qos.h> +#include <linux/sysfs.h> +#include <linux/clk.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "rt5514-spi.h" + +static struct spi_device *rt5514_spi; + +struct rt5514_dsp { + struct device *dev; + struct delayed_work copy_work; + struct mutex dma_lock; + struct snd_pcm_substream *substream; + unsigned int buf_base, buf_limit, buf_rp; + size_t buf_size; + size_t dma_offset; + size_t dsp_offset; +}; + +static const struct snd_pcm_hardware rt5514_spi_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .period_bytes_min = PAGE_SIZE, + .period_bytes_max = 0x20000 / 8, + .periods_min = 8, + .periods_max = 8, + .channels_min = 1, + .channels_max = 1, + .buffer_bytes_max = 0x20000, +}; + +static struct snd_soc_dai_driver rt5514_spi_dai = { + .name = "rt5514-dsp-cpu-dai", + .id = 0, + .capture = { + .stream_name = "DSP Capture", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}; + +static void rt5514_spi_copy_work(struct work_struct *work) +{ + struct rt5514_dsp *rt5514_dsp = + container_of(work, struct rt5514_dsp, copy_work.work); + struct snd_pcm_runtime *runtime = rt5514_dsp->substream->runtime; + size_t period_bytes, truncated_bytes = 0; + + mutex_lock(&rt5514_dsp->dma_lock); + if (!rt5514_dsp->substream) { + dev_err(rt5514_dsp->dev, "No pcm substream\n"); + goto done; + } + + period_bytes = snd_pcm_lib_period_bytes(rt5514_dsp->substream); + + if (rt5514_dsp->buf_size - rt5514_dsp->dsp_offset < period_bytes) + period_bytes = rt5514_dsp->buf_size - rt5514_dsp->dsp_offset; + + if (rt5514_dsp->buf_rp + period_bytes <= rt5514_dsp->buf_limit) { + rt5514_spi_burst_read(rt5514_dsp->buf_rp, + runtime->dma_area + rt5514_dsp->dma_offset, + period_bytes); + + if (rt5514_dsp->buf_rp + period_bytes == rt5514_dsp->buf_limit) + rt5514_dsp->buf_rp = rt5514_dsp->buf_base; + else + rt5514_dsp->buf_rp += period_bytes; + } else { + truncated_bytes = rt5514_dsp->buf_limit - rt5514_dsp->buf_rp; + rt5514_spi_burst_read(rt5514_dsp->buf_rp, + runtime->dma_area + rt5514_dsp->dma_offset, + truncated_bytes); + + rt5514_spi_burst_read(rt5514_dsp->buf_base, + runtime->dma_area + rt5514_dsp->dma_offset + + truncated_bytes, period_bytes - truncated_bytes); + + rt5514_dsp->buf_rp = rt5514_dsp->buf_base + + period_bytes - truncated_bytes; + } + + rt5514_dsp->dma_offset += period_bytes; + if (rt5514_dsp->dma_offset >= runtime->dma_bytes) + rt5514_dsp->dma_offset = 0; + + rt5514_dsp->dsp_offset += period_bytes; + + snd_pcm_period_elapsed(rt5514_dsp->substream); + + if (rt5514_dsp->dsp_offset < rt5514_dsp->buf_size) + schedule_delayed_work(&rt5514_dsp->copy_work, 5); +done: + mutex_unlock(&rt5514_dsp->dma_lock); +} + +/* PCM for streaming audio from the DSP buffer */ +static int rt5514_spi_pcm_open(struct snd_pcm_substream *substream) +{ + snd_soc_set_runtime_hwparams(substream, &rt5514_spi_pcm_hardware); + + return 0; +} + +static int rt5514_spi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct rt5514_dsp *rt5514_dsp = + snd_soc_platform_get_drvdata(rtd->platform); + int ret; + + mutex_lock(&rt5514_dsp->dma_lock); + ret = snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); + rt5514_dsp->substream = substream; + mutex_unlock(&rt5514_dsp->dma_lock); + + return ret; +} + +static int rt5514_spi_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct rt5514_dsp *rt5514_dsp = + snd_soc_platform_get_drvdata(rtd->platform); + + mutex_lock(&rt5514_dsp->dma_lock); + cancel_delayed_work_sync(&rt5514_dsp->copy_work); + rt5514_dsp->substream = NULL; + mutex_unlock(&rt5514_dsp->dma_lock); + + return snd_pcm_lib_free_vmalloc_buffer(substream); +} + +static int rt5514_spi_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct rt5514_dsp *rt5514_dsp = + snd_soc_platform_get_drvdata(rtd->platform); + u8 buf[8]; + + rt5514_dsp->dma_offset = 0; + rt5514_dsp->dsp_offset = 0; + + /** + * The address area x1800XXXX is the register address, and it cannot + * support spi burst read perfectly. So we use the spi burst read + * individually to make sure the data correctly. + */ + rt5514_spi_burst_read(RT5514_BUFFER_VOICE_BASE, (u8 *)&buf, + sizeof(buf)); + rt5514_dsp->buf_base = buf[0] | buf[1] << 8 | buf[2] << 16 | + buf[3] << 24; + + rt5514_spi_burst_read(RT5514_BUFFER_VOICE_LIMIT, (u8 *)&buf, + sizeof(buf)); + rt5514_dsp->buf_limit = buf[0] | buf[1] << 8 | buf[2] << 16 | + buf[3] << 24; + + rt5514_spi_burst_read(RT5514_BUFFER_VOICE_RP, (u8 *)&buf, + sizeof(buf)); + rt5514_dsp->buf_rp = buf[0] | buf[1] << 8 | buf[2] << 16 | + buf[3] << 24; + + rt5514_spi_burst_read(RT5514_BUFFER_VOICE_SIZE, (u8 *)&buf, + sizeof(buf)); + rt5514_dsp->buf_size = buf[0] | buf[1] << 8 | buf[2] << 16 | + buf[3] << 24; + + return 0; +} + +static int rt5514_spi_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct rt5514_dsp *rt5514_dsp = + snd_soc_platform_get_drvdata(rtd->platform); + + if (cmd == SNDRV_PCM_TRIGGER_START) { + if (rt5514_dsp->buf_base && rt5514_dsp->buf_limit && + rt5514_dsp->buf_rp && rt5514_dsp->buf_size) + schedule_delayed_work(&rt5514_dsp->copy_work, 0); + } + + return 0; +} + +static snd_pcm_uframes_t rt5514_spi_pcm_pointer( + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct rt5514_dsp *rt5514_dsp = + snd_soc_platform_get_drvdata(rtd->platform); + + return bytes_to_frames(runtime, rt5514_dsp->dma_offset); +} + +static struct snd_pcm_ops rt5514_spi_pcm_ops = { + .open = rt5514_spi_pcm_open, + .hw_params = rt5514_spi_hw_params, + .hw_free = rt5514_spi_hw_free, + .trigger = rt5514_spi_trigger, + .prepare = rt5514_spi_prepare, + .pointer = rt5514_spi_pcm_pointer, + .mmap = snd_pcm_lib_mmap_vmalloc, + .page = snd_pcm_lib_get_vmalloc_page, +}; + +static int rt5514_spi_pcm_probe(struct snd_soc_platform *platform) +{ + struct rt5514_dsp *rt5514_dsp; + + rt5514_dsp = devm_kzalloc(platform->dev, sizeof(*rt5514_dsp), + GFP_KERNEL); + + rt5514_dsp->dev = &rt5514_spi->dev; + mutex_init(&rt5514_dsp->dma_lock); + INIT_DELAYED_WORK(&rt5514_dsp->copy_work, rt5514_spi_copy_work); + snd_soc_platform_set_drvdata(platform, rt5514_dsp); + + return 0; +} + +static struct snd_soc_platform_driver rt5514_spi_platform = { + .probe = rt5514_spi_pcm_probe, + .ops = &rt5514_spi_pcm_ops, +}; + +static const struct snd_soc_component_driver rt5514_spi_dai_component = { + .name = "rt5514-spi-dai", +}; + +/** + * rt5514_spi_burst_read - Read data from SPI by rt5514 address. + * @addr: Start address. + * @rxbuf: Data Buffer for reading. + * @len: Data length, it must be a multiple of 8. + * + * + * Returns true for success. + */ +int rt5514_spi_burst_read(unsigned int addr, u8 *rxbuf, size_t len) +{ + u8 spi_cmd = RT5514_SPI_CMD_BURST_READ; + int status; + u8 write_buf[8]; + unsigned int i, end, offset = 0; + + struct spi_message message; + struct spi_transfer x[3]; + + while (offset < len) { + if (offset + RT5514_SPI_BUF_LEN <= len) + end = RT5514_SPI_BUF_LEN; + else + end = len % RT5514_SPI_BUF_LEN; + + write_buf[0] = spi_cmd; + write_buf[1] = ((addr + offset) & 0xff000000) >> 24; + write_buf[2] = ((addr + offset) & 0x00ff0000) >> 16; + write_buf[3] = ((addr + offset) & 0x0000ff00) >> 8; + write_buf[4] = ((addr + offset) & 0x000000ff) >> 0; + + spi_message_init(&message); + memset(x, 0, sizeof(x)); + + x[0].len = 5; + x[0].tx_buf = write_buf; + spi_message_add_tail(&x[0], &message); + + x[1].len = 4; + x[1].tx_buf = write_buf; + spi_message_add_tail(&x[1], &message); + + x[2].len = end; + x[2].rx_buf = rxbuf + offset; + spi_message_add_tail(&x[2], &message); + + status = spi_sync(rt5514_spi, &message); + + if (status) + return false; + + offset += RT5514_SPI_BUF_LEN; + } + + for (i = 0; i < len; i += 8) { + write_buf[0] = rxbuf[i + 0]; + write_buf[1] = rxbuf[i + 1]; + write_buf[2] = rxbuf[i + 2]; + write_buf[3] = rxbuf[i + 3]; + write_buf[4] = rxbuf[i + 4]; + write_buf[5] = rxbuf[i + 5]; + write_buf[6] = rxbuf[i + 6]; + write_buf[7] = rxbuf[i + 7]; + + rxbuf[i + 0] = write_buf[7]; + rxbuf[i + 1] = write_buf[6]; + rxbuf[i + 2] = write_buf[5]; + rxbuf[i + 3] = write_buf[4]; + rxbuf[i + 4] = write_buf[3]; + rxbuf[i + 5] = write_buf[2]; + rxbuf[i + 6] = write_buf[1]; + rxbuf[i + 7] = write_buf[0]; + } + + return true; +} + +/** + * rt5514_spi_burst_write - Write data to SPI by rt5514 address. + * @addr: Start address. + * @txbuf: Data Buffer for writng. + * @len: Data length, it must be a multiple of 8. + * + * + * Returns true for success. + */ +int rt5514_spi_burst_write(u32 addr, const u8 *txbuf, size_t len) +{ + u8 spi_cmd = RT5514_SPI_CMD_BURST_WRITE; + u8 *write_buf; + unsigned int i, end, offset = 0; + + write_buf = kmalloc(RT5514_SPI_BUF_LEN + 6, GFP_KERNEL); + + if (write_buf == NULL) + return -ENOMEM; + + while (offset < len) { + if (offset + RT5514_SPI_BUF_LEN <= len) + end = RT5514_SPI_BUF_LEN; + else + end = len % RT5514_SPI_BUF_LEN; + + write_buf[0] = spi_cmd; + write_buf[1] = ((addr + offset) & 0xff000000) >> 24; + write_buf[2] = ((addr + offset) & 0x00ff0000) >> 16; + write_buf[3] = ((addr + offset) & 0x0000ff00) >> 8; + write_buf[4] = ((addr + offset) & 0x000000ff) >> 0; + + for (i = 0; i < end; i += 8) { + write_buf[i + 12] = txbuf[offset + i + 0]; + write_buf[i + 11] = txbuf[offset + i + 1]; + write_buf[i + 10] = txbuf[offset + i + 2]; + write_buf[i + 9] = txbuf[offset + i + 3]; + write_buf[i + 8] = txbuf[offset + i + 4]; + write_buf[i + 7] = txbuf[offset + i + 5]; + write_buf[i + 6] = txbuf[offset + i + 6]; + write_buf[i + 5] = txbuf[offset + i + 7]; + } + + write_buf[end + 5] = spi_cmd; + + spi_write(rt5514_spi, write_buf, end + 6); + + offset += RT5514_SPI_BUF_LEN; + } + + kfree(write_buf); + + return 0; +} +EXPORT_SYMBOL_GPL(rt5514_spi_burst_write); + +static int rt5514_spi_probe(struct spi_device *spi) +{ + int ret; + + rt5514_spi = spi; + + ret = snd_soc_register_platform(&spi->dev, &rt5514_spi_platform); + if (ret < 0) { + dev_err(&spi->dev, "Failed to register platform.\n"); + goto err_plat; + } + + ret = snd_soc_register_component(&spi->dev, &rt5514_spi_dai_component, + &rt5514_spi_dai, 1); + if (ret < 0) { + dev_err(&spi->dev, "Failed to register component.\n"); + goto err_comp; + } + + return 0; +err_comp: + snd_soc_unregister_platform(&spi->dev); +err_plat: + + return 0; +} + +static int rt5514_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_component(&spi->dev); + snd_soc_unregister_platform(&spi->dev); + + return 0; +} + +static const struct of_device_id rt5514_of_match[] = { + { .compatible = "realtek,rt5514", }, + {}, +}; +MODULE_DEVICE_TABLE(of, rt5514_of_match); + +static struct spi_driver rt5514_spi_driver = { + .driver = { + .name = "rt5514", + .of_match_table = of_match_ptr(rt5514_of_match), + }, + .probe = rt5514_spi_probe, + .remove = rt5514_spi_remove, +}; +module_spi_driver(rt5514_spi_driver); + +MODULE_DESCRIPTION("RT5514 SPI driver"); +MODULE_AUTHOR("Oder Chiou <oder_chiou(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt5514-spi.h b/sound/soc/codecs/rt5514-spi.h new file mode 100644 index 0000000..f69b1cd --- /dev/null +++ b/sound/soc/codecs/rt5514-spi.h @@ -0,0 +1,38 @@ +/* + * rt5514-spi.h -- RT5514 driver + * + * Copyright 2015 Realtek Semiconductor Corp. + * Author: Oder Chiou <oder_chiou(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __RT5514_SPI_H__ +#define __RT5514_SPI_H__ + +/** + * RT5514_SPI_BUF_LEN is the buffer size of SPI master controller. +*/ +#define RT5514_SPI_BUF_LEN 240 + +#define RT5514_BUFFER_VOICE_BASE 0x18001034 +#define RT5514_BUFFER_VOICE_LIMIT 0x18001038 +#define RT5514_BUFFER_VOICE_RP 0x1800103c +#define RT5514_BUFFER_VOICE_SIZE 0x18001040 + +/* SPI Command */ +enum { + RT5514_SPI_CMD_16_READ = 0, + RT5514_SPI_CMD_16_WRITE, + RT5514_SPI_CMD_32_READ, + RT5514_SPI_CMD_32_WRITE, + RT5514_SPI_CMD_BURST_READ, + RT5514_SPI_CMD_BURST_WRITE, +}; + +int rt5514_spi_burst_read(unsigned int addr, u8 *rxbuf, size_t len); +int rt5514_spi_burst_write(u32 addr, const u8 *txbuf, size_t len); + +#endif /* __RT5514_SPI_H__ */ diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 879bf60..517242a 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -30,6 +30,9 @@ #include "rl6231.h" #include "rt5514.h" +#if defined(CONFIG_SND_SOC_RT5514_SPI) +#include "rt5514-spi.h" +#endif static const struct reg_sequence rt5514_i2c_patch[] = { {0x1800101c, 0x00000000}, @@ -110,6 +113,159 @@ static const struct reg_default rt5514_reg[] = { {RT5514_VENDOR_ID2, 0x10ec5514}, }; +static void rt5514_enable_dsp_prepare(struct rt5514_priv *rt5514) +{ + /* Reset */ + regmap_write(rt5514->i2c_regmap, 0x18002000, 0x000010ec); + /* LDO_I_limit */ + regmap_write(rt5514->i2c_regmap, 0x18002200, 0x00028604); + /* (for reset DSP) mini-core reset */ + regmap_write(rt5514->i2c_regmap, 0x18002f00, 0x0005514b); + /* (for reset DSP) mini-core reset */ + regmap_write(rt5514->i2c_regmap, 0x18002f00, 0x00055149); + /* DMIC_OUT1/2=HiZ */ + regmap_write(rt5514->i2c_regmap, 0x180020a0, 0x00000000); + /* PIN config */ + regmap_write(rt5514->i2c_regmap, 0x18002070, 0x00000040); + /* pll3(QN)=RCOSC*(10+2) */ + regmap_write(rt5514->i2c_regmap, 0x18002240, 0x0000000a); + /* pll3 source=RCOSC, fsi=rt_clk */ + regmap_write(rt5514->i2c_regmap, 0x18002100, 0x0000000b); + /* PU RCOSC, pll3 */ + regmap_write(rt5514->i2c_regmap, 0x18002004, 0x00808b81); + /* PD ADC1/2 */ + regmap_write(rt5514->i2c_regmap, 0x18002008, 0x00220000); + /* DSP clk source=pll3, ENABLE DSP clk */ + regmap_write(rt5514->i2c_regmap, 0x18002f08, 0x00000005); + /* 256fs=/4,DMIC_CLK_OUT=/16(disable ad2) */ + regmap_write(rt5514->i2c_regmap, 0x18002104, 0x10023541); + /* clk_sys source=mux_out */ + regmap_write(rt5514->i2c_regmap, 0x18002108, 0x00000000); + /* DSP clk= 15M*(20+1)/32 =10M */ + regmap_write(rt5514->i2c_regmap, 0x18001100, 0x00000214); + /* DB, pointer */ + regmap_write(rt5514->i2c_regmap, 0x18002148, 0x80000000); + /* DB, pop=4x */ + regmap_write(rt5514->i2c_regmap, 0x18002140, 0x3fff00fa); + /* DB, pointer */ + regmap_write(rt5514->i2c_regmap, 0x18002148, 0x00000000); + /* DFLL reset */ + regmap_write(rt5514->i2c_regmap, 0x18002124, 0x00220012); + /* DFLL, set m/n */ + regmap_write(rt5514->i2c_regmap, 0x18002110, 0x000101f4); + /* DFLL,reset DFLL */ + regmap_write(rt5514->i2c_regmap, 0x18002124, 0x80220012); + /* DFLL */ + regmap_write(rt5514->i2c_regmap, 0x18002124, 0xc0220012); + /* (I2S) i2s format, TDM 4ch */ + regmap_write(rt5514->i2c_regmap, 0x18002010, 0x10000772); + /* (I2S) source sel; tdm_0=ad0, tdm_1=ad1 */ + regmap_write(rt5514->i2c_regmap, 0x180020ac, 0x44000eee); + /* (ad0) source of DMIC */ + regmap_write(rt5514->i2c_regmap, 0x18002190, 0x0002042f); + /* (ad0) source of DMIC */ + regmap_write(rt5514->i2c_regmap, 0x18002194, 0x0002042f); + /* (ad0) DMIC-IN1 L/R select */ + regmap_write(rt5514->i2c_regmap, 0x18002198, 0x10000362); + /* (ad1) source of DMIC */ + regmap_write(rt5514->i2c_regmap, 0x180021a0, 0x0002042f); + /* (ad1) source of DMIC */ + regmap_write(rt5514->i2c_regmap, 0x180021a4, 0x0002042f); + /* (ad1) DMIC-IN2 L/R select */ + regmap_write(rt5514->i2c_regmap, 0x180021a8, 0x10000362); + /* (ad2) gain=24dB for WOV */ + regmap_write(rt5514->i2c_regmap, 0x180020d0, 0x00008a2f); + /* dsp clk auto switch */ + regmap_write(rt5514->i2c_regmap, 0x18001114, 0x00000001); + /* reduce DSP power */ + regmap_write(rt5514->i2c_regmap, 0x18001118, 0x00000001); + /* UART clk=off */ + regmap_write(rt5514->i2c_regmap, 0x18001104, 0x00000003); + /* (pitch VAD)fix1 */ + regmap_write(rt5514->i2c_regmap, 0x1800201c, 0x69f32067); + /* (pitch VAD)fix2 */ + regmap_write(rt5514->i2c_regmap, 0x18002020, 0x50d500a5); + /* (pitch VAD)fix3 */ + regmap_write(rt5514->i2c_regmap, 0x18002024, 0x000a0206); + /* (pitch VAD)fix4 */ + regmap_write(rt5514->i2c_regmap, 0x18002028, 0x01800114); + /* (hello VAD)fix1 */ + regmap_write(rt5514->i2c_regmap, 0x18002038, 0x00100010); + /* (ok VAD)fix1 */ + regmap_write(rt5514->i2c_regmap, 0x1800204c, 0x000503c8); + /* (ok VAD)fix2 */ + regmap_write(rt5514->i2c_regmap, 0x18002050, 0x001a0308); + /* (ok VAD)fix3 */ + regmap_write(rt5514->i2c_regmap, 0x18002054, 0x50020502); + /* (ok VAD)fix4 */ + regmap_write(rt5514->i2c_regmap, 0x18002058, 0x50000d18); + /* (ok VAD)fix5 */ + regmap_write(rt5514->i2c_regmap, 0x1800205c, 0x640c0b14); + /* (ok VAD)fix6 */ + regmap_write(rt5514->i2c_regmap, 0x18002060, 0x00100001); + /* (FW) SENSORY_SVSCORE(for UDT+SID) */ + regmap_write(rt5514->i2c_regmap, 0x18002fa4, 0x00000000); + /* (FW) SENSORY_THRS(for UDT+SID) */ + regmap_write(rt5514->i2c_regmap, 0x18002fa8, 0x00000000); + /* (FW) DLY_BUF_LTC_OFFSET (for ok/hello VAD) */ + regmap_write(rt5514->i2c_regmap, 0x18002fbc, 0x00000000); +} + +static void rt5514_enable_dsp(struct rt5514_priv *rt5514) +{ + /* (FW) DRIVER_CTRL0: (1)VAD timeout[7:0]: unit = 0.1s + * (2)buffer mode[15:14]: + * 0 = key phrase+voice command + * 1 = voice command + * 2 = key phrase + */ + regmap_write(rt5514->i2c_regmap, 0x18001028, 0x0000000a); + /* PU RCOSC, pll3 */ + regmap_write(rt5514->i2c_regmap, 0x18002004, 0x00808b81); + /* PD ADC1/2 */ + regmap_write(rt5514->i2c_regmap, 0x18002008, 0x00220000); + /* dsp stop */ + regmap_write(rt5514->i2c_regmap, 0x18002f00, 0x00055149); + /* CLK, 256fs=/4,DMIC_CLK_OUT=/16(enable ad2) */ + regmap_write(rt5514->i2c_regmap, 0x18002104, 0x14023541); + /* CLK, clk_sys=mux_out/1 */ + regmap_write(rt5514->i2c_regmap, 0x18002108, 0x00000000); + /* (PATH) DMIC_IN1(ri)->ad2, ad2(db_PCM)->IB2 */ + regmap_write(rt5514->i2c_regmap, 0x180020a4, 0x00808002); + /* DSP clk source=pll3, ENABLE DSP clk */ + regmap_write(rt5514->i2c_regmap, 0x18002f08, 0x00000005); + /* (opt)VAD, clr and enable pitch/hello VAD + * (0x800007d2 for ok VAD, 0x800007d3 for pitch/hello VAD) + */ + regmap_write(rt5514->i2c_regmap, 0x18002030, 0x800007d3); + /* (opt)VAD, clr and enable ok VAD + * (0x80000003 for ok VAD, 0x80000002 for pitch/hello VAD) + */ + regmap_write(rt5514->i2c_regmap, 0x18002064, 0x80000002); + /* (opt)VAD type sel(..90h:hello, ..80h:pitch, ..A0h:ok) */ + regmap_write(rt5514->i2c_regmap, 0x1800206c, 0x80000080); + /* dsp run */ + regmap_write(rt5514->i2c_regmap, 0x18002f00, 0x00055148); + /* clear IRQ */ + regmap_write(rt5514->i2c_regmap, 0x18002e04, 0x00000000); + /* (opt)VAD, release and enable pitch/hello VAD + * (0x7d2 for ok VAD, 0x7d3 for pitch/hello VAD) + */ + regmap_write(rt5514->i2c_regmap, 0x18002030, 0x000007d3); + /* (opt)VAD, release and enable ok VAD + * (0x3 for ok VAD, 0x2 for pitch/hello VAD) + */ + regmap_write(rt5514->i2c_regmap, 0x18002064, 0x00000002); + /* (opt)VAD, release and enable pitch/hello VAD + * (0x7d2 for ok VAD, 0x7d3 for pitch/hello VAD) + */ + regmap_write(rt5514->i2c_regmap, 0x18002030, 0x000007d3); + /* (opt)VAD, release and enable ok VAD + * (0x3 for ok VAD, 0x2 for pitch/hello VAD) + */ + regmap_write(rt5514->i2c_regmap, 0x18002064, 0x00000002); +} + static bool rt5514_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { @@ -248,6 +404,64 @@ static const DECLARE_TLV_DB_RANGE(bst_tlv, static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static int rt5514_dsp_voice_wake_up_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt5514_priv *rt5514 = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = rt5514->dsp_enabled; + + return 0; +} + +static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt5514_priv *rt5514 = snd_soc_component_get_drvdata(component); + struct snd_soc_codec *codec = rt5514->codec; + const struct firmware *fw = NULL; + + if (ucontrol->value.integer.value[0] == rt5514->dsp_enabled) + return 0; + + rt5514->dsp_enabled = ucontrol->value.integer.value[0]; + + if (rt5514->dsp_enabled) { + rt5514_enable_dsp_prepare(rt5514); + + request_firmware(&fw, RT5514_FIRMWARE1, codec->dev); + if (fw) { +#if defined(CONFIG_SND_SOC_RT5514_SPI) + rt5514_spi_burst_write(0x4ff60000, fw->data, + ((fw->size/8)+1)*8); +#endif + release_firmware(fw); + fw = NULL; + } + + request_firmware(&fw, RT5514_FIRMWARE2, codec->dev); + if (fw) { +#if defined(CONFIG_SND_SOC_RT5514_SPI) + rt5514_spi_burst_write(0x4ffc0000, fw->data, + ((fw->size/8)+1)*8); +#endif + release_firmware(fw); + fw = NULL; + } + + rt5514_enable_dsp(rt5514); + } else { + regcache_mark_dirty(rt5514->i2c_regmap); + regcache_mark_dirty(rt5514->regmap); + regcache_sync(rt5514->i2c_regmap); + regcache_sync(rt5514->regmap); + } + + return 0; +} + static const struct snd_kcontrol_new rt5514_snd_controls[] = { SOC_DOUBLE_TLV("MIC Boost Volume", RT5514_ANA_CTRL_MICBST, RT5514_SEL_BSTL_SFT, RT5514_SEL_BSTR_SFT, 8, 0, bst_tlv), @@ -257,6 +471,8 @@ static const struct snd_kcontrol_new rt5514_snd_controls[] = { SOC_DOUBLE_R_TLV("ADC2 Capture Volume", RT5514_DOWNFILTER1_CTRL1, RT5514_DOWNFILTER1_CTRL2, RT5514_AD_GAIN_SFT, 127, 0, adc_vol_tlv), + SOC_SINGLE_EXT("DSP Voice Wake Up", SND_SOC_NOPM, 0, 1, 0, + rt5514_dsp_voice_wake_up_get, rt5514_dsp_voice_wake_up_put), }; /* ADC Mixer*/ @@ -799,6 +1015,38 @@ static int rt5514_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, return 0; } +static int rt5514_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct rt5514_priv *rt5514 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (snd_soc_codec_get_bias_level(codec) == + SND_SOC_BIAS_STANDBY) { + /** + * If the DSP is enabled in start of recording, the + * DSP should be disabled, and sync back to normal + * recording settings to make sure recording + * properly. + */ + if (rt5514->dsp_enabled) { + rt5514->dsp_enabled = 0; + regcache_mark_dirty(rt5514->i2c_regmap); + regcache_mark_dirty(rt5514->regmap); + regcache_sync(rt5514->i2c_regmap); + regcache_sync(rt5514->regmap); + } + } + break; + + default: + break; + } + + return 0; +} + static int rt5514_probe(struct snd_soc_codec *codec) { struct rt5514_priv *rt5514 = snd_soc_codec_get_drvdata(codec); @@ -857,6 +1105,7 @@ struct snd_soc_dai_driver rt5514_dai[] = { static struct snd_soc_codec_driver soc_codec_dev_rt5514 = { .probe = rt5514_probe, + .set_bias_level = rt5514_set_bias_level, .idle_bias_off = true, .controls = rt5514_snd_controls, .num_controls = ARRAY_SIZE(rt5514_snd_controls), diff --git a/sound/soc/codecs/rt5514.h b/sound/soc/codecs/rt5514.h index 6ad8a61..6e89e7d 100644 --- a/sound/soc/codecs/rt5514.h +++ b/sound/soc/codecs/rt5514.h @@ -225,6 +225,9 @@ #define RT5514_PLL_INP_MAX 40000000 #define RT5514_PLL_INP_MIN 256000 +#define RT5514_FIRMWARE1 "rt5514_dsp_fw1.bin" +#define RT5514_FIRMWARE2 "rt5514_dsp_fw2.bin" + /* System Clock Source */ enum { RT5514_SCLK_S_MCLK, @@ -247,6 +250,7 @@ struct rt5514_priv { int pll_src; int pll_in; int pll_out; + int dsp_enabled; }; #endif /* __RT5514_H__ */ -- 1.8.1.1.439.g50a6b54

2 7