December 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] sound: hda_intel: add card number to irq description
by Heiner Kallweit 22 Dec '15

22 Dec '15

Currently the info in /proc/interrupts doesn't allow to figure out which interrupt belongs to which card (HDMI, PCH, ..). Therefore add card details to the interrupt description. With the patch the info in /proc/interrupts looks like this: PCI-MSI 442368-edge snd_hda_intel:card1 PCI-MSI 49152-edge snd_hda_intel:card0 This could be partially reused for the hda_tegra driver as it also uses struct azx. Signed-off-by: Heiner Kallweit <hkallweit1(a)gmail.com> --- sound/pci/hda/hda_controller.h | 2 ++ sound/pci/hda/hda_intel.c | 4 +++- 2 files changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index ec63bbf..9c6344f 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -125,6 +125,8 @@ struct azx { int num_streams; const int *jackpoll_ms; /* per-card jack poll interval */ + char irq_descr[32]; /* Interrupt description */ + /* Register interaction. */ const struct hda_controller_ops *ops; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 83800ac..ef4e06b 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -725,7 +725,7 @@ static int azx_acquire_irq(struct azx *chip, int do_disconnect) if (request_irq(chip->pci->irq, azx_interrupt, chip->msi ? 0 : IRQF_SHARED, - KBUILD_MODNAME, chip)) { + chip->irq_descr, chip)) { dev_err(chip->card->dev, "unable to grab IRQ %d, disabling device\n", chip->pci->irq); @@ -1605,6 +1605,8 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, check_msi(chip); chip->dev_index = dev; chip->jackpoll_ms = jackpoll_ms; + snprintf(chip->irq_descr, sizeof(chip->irq_descr), "%s:%s", + KBUILD_MODNAME, dev_name(&card->card_dev)); INIT_LIST_HEAD(&chip->pcm_list); INIT_WORK(&hda->irq_pending_work, azx_irq_pending_work); INIT_LIST_HEAD(&hda->list); -- 2.6.4

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[alsa-devel] [RFC PATCH 0/2] ALSA: hda - DP MST audio for Jack support
by libin.yang＠linux.intel.com 22 Dec '15

22 Dec '15

From: Libin Yang <libin.yang(a)linux.intel.com> This is the patch set for Jack support for DP MST audio Libin Yang (2): ALSA: hda - jack support DP MST audio ALSA: hda - hdmi audio add dynamically jack binding to pin sound/pci/hda/hda_jack.c | 186 ++++++++++++++++++++++++++++++++++++++++----- sound/pci/hda/hda_jack.h | 32 +++++++- sound/pci/hda/patch_hdmi.c | 116 ++++++++++++++++++++++------ 3 files changed, 288 insertions(+), 46 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH 0/8] ALSA: oxfw: obsolete snd-scs1x with snd-oxfw
by Takashi Sakamoto 22 Dec '15

22 Dec '15

Hi, This patchset update a part of my previous RFCv2 (patch 10-17). [alsa-devel] [RFC][PATCH 00/17 v2] ALSA: oxfw: refactoring and merging scs1x module http://mailman.alsa-project.org/pipermail/alsa-devel/2015-December/101558.h… This patchset should be accumurated to the other patches under reviewing. [alsa-devel] [PATCH 0/4] ALSA: oxfw: localize stuffs of old firewire-speakers module http://mailman.alsa-project.org/pipermail/alsa-devel/2015-December/102071.h… This patchset obsoletes snd-scs1x module with snd-oxfw module because the models supported by snd-scs1x module are based on OXFW971, which snd-oxfw supports, too. So snd-oxfw should support them. As a result, one of the models gets packet streaming functionality to receive PCM samples. This is a lack of functionalities in snd-scs1x. Usually I won't post additional patches during the others are under reviewing. But it's near the end of year. I'd like to post these patches in advance to avoid year-end busyness. I'm happy if you review them and these patches go forward, then I can get a short rest in the year-end. I note that I have the other patches for ALSA dice module, too. I'm also happy to start reviewing process for them after this patchset goes forward... Changes from RFCv2: * remove sound/firewire/scs1x.c (I forget it) * revise operations in rawmidi callbacks Takashi Sakamoto (8): ALSA: oxfw: add scs1x layer ALSA: oxfw: allocate own address region for SCS.1 series ALSA: oxfw: copy handlers of asynchronous transaction for MIDI capture ALSA: oxfw: add MIDI capture port for SCS.1 models ALSA: oxfw: copy handlers of asynchronous transaction for MIDI playback ALSA: oxfw: add MIDI playback port for SCS.1 models ALSA: oxfw: obsolete scs1x module ALSA: oxfw: add stream format quirk for SCS.1 models sound/firewire/Kconfig | 12 +- sound/firewire/Makefile | 2 - sound/firewire/oxfw/Makefile | 2 +- sound/firewire/oxfw/oxfw-scs1x.c | 406 ++++++++++++++++++++++++++++++ sound/firewire/oxfw/oxfw.c | 38 ++- sound/firewire/oxfw/oxfw.h | 2 + sound/firewire/scs1x.c | 530 --------------------------------------- 7 files changed, 446 insertions(+), 546 deletions(-) create mode 100644 sound/firewire/oxfw/oxfw-scs1x.c delete mode 100644 sound/firewire/scs1x.c -- 2.5.0

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[alsa-devel] [PATCH] ALSA: usb-audio: use list_for_each_entry_continue_reverse
by Geliang Tang 22 Dec '15

22 Dec '15

For better readability, use list_for_each_entry_continue_reverse() in have_dup_chmap(). Signed-off-by: Geliang Tang <geliangtang(a)163.com> --- sound/usb/stream.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 8ee14f2..c4dc577 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -125,11 +125,9 @@ static int usb_chmap_ctl_info(struct snd_kcontrol *kcontrol, static bool have_dup_chmap(struct snd_usb_substream *subs, struct audioformat *fp) { - struct list_head *p; + struct audioformat *prev = fp; - for (p = fp->list.prev; p != &subs->fmt_list; p = p->prev) { - struct audioformat *prev; - prev = list_entry(p, struct audioformat, list); + list_for_each_entry_continue_reverse(prev, &subs->fmt_list, list) { if (prev->chmap && !memcmp(prev->chmap, fp->chmap, sizeof(*fp->chmap))) return true; -- 2.5.0

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[alsa-devel] [PATCH] ALSA: hda/realtek - Fix silent headphone output on MacPro 4, 1 (v2)
by Mario Kleiner 22 Dec '15

22 Dec '15

Without this patch, internal speaker and line-out work, but front headphone output jack stays silent on the Mac Pro 4,1. This code path also gets executed on the MacPro 5,1 due to identical codec SSID, but i don't know if it has any positive or adverse effects there or not. (v2) Implement feedback from Takashi Iwai: Reuse alc889_fixup_mbp_vref and just add a new nid 0x19 for the MacPro 4,1. Signed-off-by: Mario Kleiner <mario.kleiner.de(a)gmail.com> Cc: <stable(a)vger.kernel.org> --- sound/pci/hda/patch_realtek.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9bedf7c..0e654e4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1755,6 +1755,7 @@ enum { ALC889_FIXUP_MBA11_VREF, ALC889_FIXUP_MBA21_VREF, ALC889_FIXUP_MP11_VREF, + ALC889_FIXUP_MP41_VREF, ALC882_FIXUP_INV_DMIC, ALC882_FIXUP_NO_PRIMARY_HP, ALC887_FIXUP_ASUS_BASS, @@ -1843,7 +1844,7 @@ static void alc889_fixup_mbp_vref(struct hda_codec *codec, const struct hda_fixup *fix, int action) { struct alc_spec *spec = codec->spec; - static hda_nid_t nids[2] = { 0x14, 0x15 }; + static hda_nid_t nids[3] = { 0x14, 0x15, 0x19 }; int i; if (action != HDA_FIXUP_ACT_INIT) @@ -2133,6 +2134,12 @@ static const struct hda_fixup alc882_fixups[] = { .chained = true, .chain_id = ALC885_FIXUP_MACPRO_GPIO, }, + [ALC889_FIXUP_MP41_VREF] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc889_fixup_mbp_vref, + .chained = true, + .chain_id = ALC885_FIXUP_MACPRO_GPIO, + }, [ALC882_FIXUP_INV_DMIC] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_inv_dmic, @@ -2215,7 +2222,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4100, "Macmini 3,1", ALC889_FIXUP_IMAC91_VREF), - SND_PCI_QUIRK(0x106b, 0x4200, "Mac Pro 5,1", ALC885_FIXUP_MACPRO_GPIO), + SND_PCI_QUIRK(0x106b, 0x4200, "Mac Pro 4,1/5,1", ALC889_FIXUP_MP41_VREF), SND_PCI_QUIRK(0x106b, 0x4300, "iMac 9,1", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4900, "iMac 9,1 Aluminum", ALC889_FIXUP_IMAC91_VREF), -- 1.9.1

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Re: [alsa-devel] [PATCH v1 2/2] dt-bindings: sound: add devicetree document for rt5616
by Caesar Wang 22 Dec '15

22 Dec '15

Hi 在 2015年12月22日 05:15, Frank Rowand 写道: > On 12/21/2015 6:56 AM, Caesar Wang wrote: >> Add the description for rt5616 codec. >> >> Signed-off-by: Caesar Wang <wxt(a)rock-chips.com> >> >> --- >> >> Changes in v1: >> - As Heiko comments, remove the not exist option properties. >> >> Documentation/devicetree/bindings/sound/rt5616.txt | 36 ++++++++++++++++++++++ >> 1 file changed, 36 insertions(+) >> create mode 100644 Documentation/devicetree/bindings/sound/rt5616.txt >> >> diff --git a/Documentation/devicetree/bindings/sound/rt5616.txt b/Documentation/devicetree/bindings/sound/rt5616.txt >> new file mode 100644 >> index 0000000..2030a22 >> --- /dev/null >> +++ b/Documentation/devicetree/bindings/sound/rt5616.txt >> @@ -0,0 +1,36 @@ >> +RT5616 audio CODEC >> + >> +This device supports I2C only. >> + >> +Required properties: >> + >> +- compatible : "realtek,rt5616". >> + >> +- reg : The I2C address of the device. >> + >> +Pins on the device (for linking into audio routes) for RT5616: >> + >> + * IN1P >> + * IN2P >> + * IN2N >> + * LOUTL >> + * LOUTR >> + * CPN2 >> + * CPP2 >> + * CPN1 >> + * CPP1 >> + * HPO_R >> + * HPO_L >> + * ADCDAT1 >> + * DACDAT1 >> + * LRCK1 >> + * BCLK1 >> + * MCLK >> + * GPIO1 >> + >> +Example: >> + >> +rt5616 { > The node name should be rt5616@1b (including the reg value > from the node). In general, that's right. Okay, fixed in next version. > >> + compatible = "realtek,rt5616"; >> + reg = <0x1b>; >> +}; >> > > > -- caesar wang | software engineer | wxt(a)rock-chip.com

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[alsa-devel] [PATCH] ALSA: hda/realtek - Fix silent headphone output on MacPro 4, 1
by Mario Kleiner 22 Dec '15

22 Dec '15

Without this patch, internal speaker and line-out work, but front headphone output jack stays silent on the Mac Pro 4,1. This code path also gets executed on the MacPro 5,1 due to identical codec SSID, but i don't know if it has any positive or adverse effects there or not. Signed-off-by: Mario Kleiner <mario.kleiner.de(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 32 +++++++++++++++++++++++++++++++- 1 file changed, 31 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9bedf7c..6224bdd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1755,6 +1755,7 @@ enum { ALC889_FIXUP_MBA11_VREF, ALC889_FIXUP_MBA21_VREF, ALC889_FIXUP_MP11_VREF, + ALC889_FIXUP_MP41_VREF, ALC882_FIXUP_INV_DMIC, ALC882_FIXUP_NO_PRIMARY_HP, ALC887_FIXUP_ASUS_BASS, @@ -1905,6 +1906,29 @@ static void alc889_fixup_mba21_vref(struct hda_codec *codec, alc889_fixup_mac_pins(codec, nids, ARRAY_SIZE(nids)); } +/* Set VREF on HP pin on Mac Pro 4,1 */ +static void alc889_fixup_mp41_vref(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + static hda_nid_t nids[1] = { 0x19 }; + int i; + + if (action != HDA_FIXUP_ACT_INIT) + return; + + for (i = 0; i < ARRAY_SIZE(nids); i++) { + unsigned int val = snd_hda_codec_get_pincfg(codec, nids[i]); + if (get_defcfg_device(val) != AC_JACK_HP_OUT) + continue; + val = snd_hda_codec_get_pin_target(codec, nids[i]); + val |= AC_PINCTL_VREF_80; + snd_hda_set_pin_ctl(codec, nids[i], val); + spec->gen.keep_vref_in_automute = 1; + break; + } +} + /* Don't take HP output as primary * Strangely, the speaker output doesn't work on Vaio Z and some Vaio * all-in-one desktop PCs (for example VGC-LN51JGB) through DAC 0x05 @@ -2133,6 +2157,12 @@ static const struct hda_fixup alc882_fixups[] = { .chained = true, .chain_id = ALC885_FIXUP_MACPRO_GPIO, }, + [ALC889_FIXUP_MP41_VREF] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc889_fixup_mp41_vref, + .chained = true, + .chain_id = ALC885_FIXUP_MACPRO_GPIO, + }, [ALC882_FIXUP_INV_DMIC] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_inv_dmic, @@ -2215,7 +2245,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4100, "Macmini 3,1", ALC889_FIXUP_IMAC91_VREF), - SND_PCI_QUIRK(0x106b, 0x4200, "Mac Pro 5,1", ALC885_FIXUP_MACPRO_GPIO), + SND_PCI_QUIRK(0x106b, 0x4200, "Mac Pro 4,1/5,1", ALC889_FIXUP_MP41_VREF), SND_PCI_QUIRK(0x106b, 0x4300, "iMac 9,1", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4900, "iMac 9,1 Aluminum", ALC889_FIXUP_IMAC91_VREF), -- 1.9.1

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[alsa-devel] [PATCH 00/13] Add ASoC support for AMD APUs [v5]
by Alex Deucher 21 Dec '15

21 Dec '15

This patch set implements support for i2s audio and new AMD GPUs. The i2s codec is fed by a DMA engine on the GPU. To handle this we create mfd cells which we hang the i2s codec and DMA engine on. Because of this, this patch set covers two subsystems: drm and alsa. The drm patches add support for the ACP hw block which provides the DMA engine for the i2s codec. The alsa patches add the ASoC driver for the i2s codec. Since the alsa changes depend on the drm changes, I'd like to take the alsa patches in via the drm tree. V2 changes: - Use the MFD subsystem rather than adding our own bus - Squash all sub-feature patches together - fix comments mentioned in previous review V3 changes: - Update the designware driver to handle slave mode, amd specific features - Use the designware driver directly for i2s - Move the DMA handling from the GPU driver into the AMD ASoC driver - Change the license on the ASoC driver to GPL v4 changes: - patch "ASoC : dwc : support dw i2s in slave mode" accepted - Add a _dai_fmt() operation that checks to make sure that the mode we're setting corresponds to what we read back from the hardware. - Split specific quirks into separate patches - Set the specific quirks that apply to AMD chips in the acp driver v5 changes: - patch "ASoC : dwc : add check for master/slave format" accepted - Fix MFD_CORE selection in ACP Kconfig - Add irq domain support to amdgpu driver - Use genirq in ACP DMA driver - Export some genpd symbols to support ACP powergating (Acked by PM maintainer) - Use genpd for ACP powergating - add separate capture and playback instances of dws in ACP init - add runtime suspend support for dws in master mode Patch 9 adds the register headers for the ACP block which is a pretty big patch so I've excluded it from email. The entire patch set can be viewed here: http://cgit.freedesktop.org/~agd5f/linux/log/?h=acp-upstream8 Thanks, Alex Alex Deucher (3): drm/amdgpu/cgs: add an interface to access PCI resources drm/amdgpu: add irq domain support drm/amd: add pm domain for ACP IP sub blocks Maruthi Bayyavarapu (1): drm/amd: add ACP driver support Maruthi Srinivas Bayyavarapu (9): ASoC: dwc: add runtime suspend/resume functionality ASoC: dwc: add quirk for different register offset ASoC: dwc: reconfigure dwc in 'resume' from 'suspend' PM / Domains: export symbols to add/remove devices from genpd ASoC : AMD : add ACP 2.2 register headers ASoC: AMD: add ACP 2.x IP DMA abstraction layer ASoC: AMD: add AMD ASoC ACP 2.x DMA driver ASoC: AMD: add pm ops ASoC: AMD: Manage ACP 2.x SRAM banks power drivers/base/power/domain.c | 2 + drivers/gpu/drm/Kconfig | 2 + drivers/gpu/drm/amd/acp/Kconfig | 11 + drivers/gpu/drm/amd/acp/Makefile | 9 + drivers/gpu/drm/amd/acp/acp_hw.c | 50 + drivers/gpu/drm/amd/acp/include/acp_gfx_if.h | 34 + drivers/gpu/drm/amd/amdgpu/Makefile | 13 +- drivers/gpu/drm/amd/amdgpu/amdgpu.h | 12 + drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c | 501 ++++++ drivers/gpu/drm/amd/amdgpu/amdgpu_acp.h | 42 + drivers/gpu/drm/amd/amdgpu/amdgpu_cgs.c | 36 + drivers/gpu/drm/amd/amdgpu/amdgpu_irq.c | 108 +- drivers/gpu/drm/amd/amdgpu/amdgpu_irq.h | 9 + drivers/gpu/drm/amd/amdgpu/cik_ih.c | 6 + drivers/gpu/drm/amd/amdgpu/cz_ih.c | 7 + drivers/gpu/drm/amd/amdgpu/iceland_ih.c | 7 + drivers/gpu/drm/amd/amdgpu/tonga_ih.c | 7 + drivers/gpu/drm/amd/amdgpu/vi.c | 12 + drivers/gpu/drm/amd/include/amd_shared.h | 1 + drivers/gpu/drm/amd/include/cgs_common.h | 34 + include/sound/designware_i2s.h | 5 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/amd/Kconfig | 4 + sound/soc/amd/Makefile | 3 + sound/soc/amd/acp-pcm-dma.c | 560 +++++++ sound/soc/amd/acp.c | 666 ++++++++ sound/soc/amd/acp.h | 140 ++ sound/soc/amd/include/acp_2_2_d.h | 609 +++++++ sound/soc/amd/include/acp_2_2_enum.h | 1068 ++++++++++++ sound/soc/amd/include/acp_2_2_sh_mask.h | 2292 ++++++++++++++++++++++++++ sound/soc/dwc/designware_i2s.c | 113 +- 32 files changed, 6325 insertions(+), 40 deletions(-) create mode 100644 drivers/gpu/drm/amd/acp/Kconfig create mode 100644 drivers/gpu/drm/amd/acp/Makefile create mode 100644 drivers/gpu/drm/amd/acp/acp_hw.c create mode 100644 drivers/gpu/drm/amd/acp/include/acp_gfx_if.h create mode 100644 drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c create mode 100644 drivers/gpu/drm/amd/amdgpu/amdgpu_acp.h create mode 100644 sound/soc/amd/Kconfig create mode 100644 sound/soc/amd/Makefile create mode 100644 sound/soc/amd/acp-pcm-dma.c create mode 100644 sound/soc/amd/acp.c create mode 100644 sound/soc/amd/acp.h create mode 100644 sound/soc/amd/include/acp_2_2_d.h create mode 100644 sound/soc/amd/include/acp_2_2_enum.h create mode 100644 sound/soc/amd/include/acp_2_2_sh_mask.h -- 1.8.3.1

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[alsa-devel] [PATCH v1 1/5] ALSA: fm801: propagate TUNER_ONLY bit when autodetected
by Andy Shevchenko 21 Dec '15

21 Dec '15

The commit d7ba858a7f7a (ALSA: fm801: implement TEA575x tuner autodetection) brings autodetection to the driver. However the autodetection algorithm misses the TUNER_ONLY bit if it is supplied by the user. Thus, user gets weird messages and no card registered. snd_fm801 0000:0d:01.0: detected TEA575x radio type SF64-PCR snd_fm801 0000:0d:01.0: AC'97 interface is busy (1) snd_fm801 0000:0d:01.0: AC'97 interface is busy (1) ... snd_fm801 0000:0d:01.0: AC'97 0 does not respond - RESET snd_fm801 0000:0d:01.0: AC'97 interface is busy (1) snd_fm801 0000:0d:01.0: AC'97 interface is busy (1) snd_fm801 0000:0d:01.0: AC'97 0 access is not valid [0x0], removing mixer. snd_fm801: probe of 0000:0d:01.0 failed with error -5 Do a copy of TUNER_ONLY bit to be applied after autodetection is done. Fixes: d7ba858a7f7a (ALSA: fm801: implement TEA575x tuner autodetection) Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Cc: Ondrej Zary <linux(a)rainbow-software.org> --- sound/pci/fm801.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 4c9d804..6114d81 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -1269,6 +1269,8 @@ static int snd_fm801_create(struct snd_card *card, return -ENODEV; } } else if ((tea575x_tuner & TUNER_TYPE_MASK) == 0) { + unsigned int tuner_only = tea575x_tuner & TUNER_ONLY; + /* autodetect tuner connection */ for (tea575x_tuner = 1; tea575x_tuner <= 3; tea575x_tuner++) { chip->tea575x_tuner = tea575x_tuner; @@ -1283,6 +1285,8 @@ static int snd_fm801_create(struct snd_card *card, dev_err(card->dev, "TEA575x radio not found\n"); chip->tea575x_tuner = TUNER_DISABLED; } + + chip->tea575x_tuner |= tuner_only; } if (!(chip->tea575x_tuner & TUNER_DISABLED)) { strlcpy(chip->tea.card, get_tea575x_gpio(chip)->name, -- 2.6.4

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[alsa-devel] [PATCH v6] sun4i-codec: Add FM, Line and Mic inputs
by Danny Milosavljevic 21 Dec '15

21 Dec '15

Add inputs to sun4i-codec: - FM-in Left and Right - Line-in Left and Right - Mic1-in - Mic2-in Signed-off-by: Danny Milosavljevic <dannym+a(a)scratchpost.org> Tested-by: Danny Milosavljevic <dannym+a(a)scratchpost.org> (ONLY ON A20!) --- Hi, this is the sixth version of the patch that adds inputs to sun4i-codec. Changes compared to v5 are: - Mic preamplifier special cases for A20 and A10 now are now not icky: There are two different _widget arrays and the probe() function now selects the right one to pass to snd_soc_register_codec() unmodified. - sun7i-specific things (A20-specific things) are now grouped together. I successfully tested it again on an A20 board using alsamixer, headphones, a radio and my ears. Note that because of missing capturing support I tested only the mixing, for Mic, Line, and FM. The patches are on top of <git://git.kernel.org/pub/scm/linux/kernel/git/mripard/linux.git>, branch "sunxi/for-next". Regards, Danny --- b/sound/soc/sunxi/sun4i-codec.c | 179 +++++++++++++++++++++++++++++++++++++--- 1 file changed, 166 insertions(+), 13 deletions(-) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index bcbf4da..4241999 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -57,9 +57,20 @@ #define SUN4I_CODEC_DAC_ACTL_DACAENR (31) #define SUN4I_CODEC_DAC_ACTL_DACAENL (30) #define SUN4I_CODEC_DAC_ACTL_MIXEN (29) +#define SUN4I_CODEC_DAC_ACTL_LNG (26) +#define SUN4I_CODEC_DAC_ACTL_FMG (23) +#define SUN4I_CODEC_DAC_ACTL_MICG (20) +#define SUN4I_CODEC_DAC_ACTL_LLNS (19) +#define SUN4I_CODEC_DAC_ACTL_RLNS (18) +#define SUN4I_CODEC_DAC_ACTL_LFMS (17) +#define SUN4I_CODEC_DAC_ACTL_RFMS (16) #define SUN4I_CODEC_DAC_ACTL_LDACLMIXS (15) #define SUN4I_CODEC_DAC_ACTL_RDACRMIXS (14) #define SUN4I_CODEC_DAC_ACTL_LDACRMIXS (13) +#define SUN4I_CODEC_DAC_ACTL_MIC1LS (12) +#define SUN4I_CODEC_DAC_ACTL_MIC1RS (11) +#define SUN4I_CODEC_DAC_ACTL_MIC2LS (10) +#define SUN4I_CODEC_DAC_ACTL_MIC2RS (9) #define SUN4I_CODEC_DAC_ACTL_DACPAS (8) #define SUN4I_CODEC_DAC_ACTL_MIXPAS (7) #define SUN4I_CODEC_DAC_ACTL_PA_MUTE (6) @@ -84,8 +95,11 @@ #define SUN4I_CODEC_ADC_ACTL_PREG1EN (29) #define SUN4I_CODEC_ADC_ACTL_PREG2EN (28) #define SUN4I_CODEC_ADC_ACTL_VMICEN (27) +#define SUN4I_CODEC_ADC_ACTL_PREG1_A10 (25) +#define SUN4I_CODEC_ADC_ACTL_PREG2_A10 (23) #define SUN4I_CODEC_ADC_ACTL_VADCG (20) #define SUN4I_CODEC_ADC_ACTL_ADCIS (17) +#define SUN4I_CODEC_ADC_ACTL_LNRDF (16) #define SUN4I_CODEC_ADC_ACTL_PA_EN (4) #define SUN4I_CODEC_ADC_ACTL_DDE (3) #define SUN4I_CODEC_ADC_DEBUG (0x2c) @@ -94,7 +108,6 @@ #define SUN4I_CODEC_DAC_TXCNT (0x30) #define SUN4I_CODEC_ADC_RXCNT (0x34) #define SUN4I_CODEC_AC_SYS_VERI (0x38) -#define SUN4I_CODEC_AC_MIC_PHONE_CAL (0x3c) struct sun4i_codec { struct device *dev; @@ -402,16 +415,77 @@ static const struct snd_kcontrol_new sun4i_codec_pa_mute = SUN4I_CODEC_DAC_ACTL_PA_MUTE, 1, 0); static DECLARE_TLV_DB_SCALE(sun4i_codec_pa_volume_scale, -6300, 100, 1); +static DECLARE_TLV_DB_SCALE(sun4i_codec_linein_loopback_gain_scale, -150, 150, 0); +static DECLARE_TLV_DB_SCALE(sun4i_codec_fmin_loopback_gain_scale, -450, 150, 0); +static DECLARE_TLV_DB_SCALE(sun4i_codec_micin_loopback_gain_scale, -450, 150, 0); +static DECLARE_TLV_DB_RANGE(sun4i_codec_micin_preamp_gain_scale_a10, + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 7, TLV_DB_SCALE_ITEM(3500, 300, 0) +); + +#define SUN4I_COMMON_CODEC_WIDGETS \ + SOC_SINGLE_TLV("PA Volume", SUN4I_CODEC_DAC_ACTL, \ + SUN4I_CODEC_DAC_ACTL_PA_VOL, 0x3F, 0, \ + sun4i_codec_pa_volume_scale), \ + /* Line-In, FM-In, Mic1-In, Mic2-In */ \ + SOC_SINGLE_TLV("Line-In Playback Gain", \ + SUN4I_CODEC_DAC_ACTL, \ + SUN4I_CODEC_DAC_ACTL_LNG, \ + 1, \ + 0, \ + sun4i_codec_linein_loopback_gain_scale), \ + SOC_SINGLE_TLV("FM-In Playback Gain", \ + SUN4I_CODEC_DAC_ACTL, \ + SUN4I_CODEC_DAC_ACTL_FMG, \ + 3, \ + 0, \ + sun4i_codec_fmin_loopback_gain_scale), \ + SOC_SINGLE_TLV("Mic-In Playback Gain", \ + SUN4I_CODEC_DAC_ACTL, \ + SUN4I_CODEC_DAC_ACTL_MICG, \ + 7, \ + 0, \ + sun4i_codec_micin_loopback_gain_scale), \ + SOC_SINGLE_TLV("Mic1-In Boost Switch", \ + SUN4I_CODEC_ADC_ACTL, \ + SUN4I_CODEC_ADC_ACTL_PREG1EN, \ + 1, \ + 0, \ + NULL), \ + SOC_SINGLE_TLV("Mic2-In Boost Switch", \ + SUN4I_CODEC_ADC_ACTL, \ + SUN4I_CODEC_ADC_ACTL_PREG2EN, \ + 1, \ + 0, \ + NULL) static const struct snd_kcontrol_new sun4i_codec_widgets[] = { - SOC_SINGLE_TLV("PA Volume", SUN4I_CODEC_DAC_ACTL, - SUN4I_CODEC_DAC_ACTL_PA_VOL, 0x3F, 0, - sun4i_codec_pa_volume_scale), + SUN4I_COMMON_CODEC_WIDGETS, + SOC_SINGLE_TLV("Mic1-In Boost Gain", + SUN4I_CODEC_ADC_ACTL, + SUN4I_CODEC_ADC_ACTL_PREG1_A10, + 3, + 0, + sun4i_codec_micin_preamp_gain_scale_a10), + SOC_SINGLE_TLV("Mic2-In Boost Gain", + SUN4I_CODEC_ADC_ACTL, + SUN4I_CODEC_ADC_ACTL_PREG2_A10, + 3, + 0, + sun4i_codec_micin_preamp_gain_scale_a10), }; static const struct snd_kcontrol_new sun4i_codec_left_mixer_controls[] = { SOC_DAPM_SINGLE("Left DAC Playback Switch", SUN4I_CODEC_DAC_ACTL, SUN4I_CODEC_DAC_ACTL_LDACLMIXS, 1, 0), + SOC_DAPM_SINGLE("Left Line-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_LLNS, 1, 0), + SOC_DAPM_SINGLE("Left FM-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_LFMS, 1, 0), + SOC_DAPM_SINGLE("Mic1-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_MIC1LS, 1, 0), + SOC_DAPM_SINGLE("Mic2-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_MIC2LS, 1, 0), }; static const struct snd_kcontrol_new sun4i_codec_right_mixer_controls[] = { @@ -419,6 +493,14 @@ static const struct snd_kcontrol_new sun4i_codec_right_mixer_controls[] = { SUN4I_CODEC_DAC_ACTL_RDACRMIXS, 1, 0), SOC_DAPM_SINGLE("Left DAC Playback Switch", SUN4I_CODEC_DAC_ACTL, SUN4I_CODEC_DAC_ACTL_LDACRMIXS, 1, 0), + SOC_DAPM_SINGLE("Right Line-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_RLNS, 1, 0), + SOC_DAPM_SINGLE("Right FM-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_RFMS, 1, 0), + SOC_DAPM_SINGLE("Mic1-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_MIC1RS, 1, 0), + SOC_DAPM_SINGLE("Mic2-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_MIC2RS, 1, 0), }; static const struct snd_kcontrol_new sun4i_codec_pa_mixer_controls[] = { @@ -462,8 +544,16 @@ static const struct snd_soc_dapm_widget sun4i_codec_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HP Right"), SND_SOC_DAPM_OUTPUT("HP Left"), + + SND_SOC_DAPM_INPUT("Line-In Right"), + SND_SOC_DAPM_INPUT("Line-In Left"), + SND_SOC_DAPM_INPUT("FM-In Right"), + SND_SOC_DAPM_INPUT("FM-In Left"), + SND_SOC_DAPM_INPUT("Mic1-In"), + SND_SOC_DAPM_INPUT("Mic2-In"), }; +/* {sink, control, source} */ static const struct snd_soc_dapm_route sun4i_codec_dapm_routes[] = { /* Left DAC Routes */ { "Left DAC", NULL, "DAC" }, @@ -490,9 +580,23 @@ static const struct snd_soc_dapm_route sun4i_codec_dapm_routes[] = { { "Pre-Amplifier Mute", "Switch", "Pre-Amplifier" }, { "HP Right", NULL, "Pre-Amplifier Mute" }, { "HP Left", NULL, "Pre-Amplifier Mute" }, + + /* Line-In, FM-In */ + { "Right Mixer", "Right Line-In Playback Switch", "Line-In Right" }, + { "Left Mixer", "Left Line-In Playback Switch", "Line-In Left" }, + { "Right Mixer", "Right FM-In Playback Switch", "FM-In Right" }, + { "Left Mixer", "Left FM-In Playback Switch", "FM-In Left" }, + + /* Mic1-In */ + { "Right Mixer", "Mic1-In Playback Switch", "Mic1-In" }, + { "Left Mixer", "Mic1-In Playback Switch", "Mic1-In" }, + + /* Mic2-In */ + { "Right Mixer", "Mic2-In Playback Switch", "Mic2-In" }, + { "Left Mixer", "Mic2-In Playback Switch", "Mic2-In" }, }; -static struct snd_soc_codec_driver sun4i_codec_codec = { +static const struct snd_soc_codec_driver sun4i_codec_codec = { .controls = sun4i_codec_widgets, .num_controls = ARRAY_SIZE(sun4i_codec_widgets), .dapm_widgets = sun4i_codec_dapm_widgets, @@ -533,13 +637,6 @@ static struct snd_soc_dai_driver dummy_cpu_dai = { }, }; -static const struct regmap_config sun4i_codec_regmap_config = { - .reg_bits = 32, - .reg_stride = 4, - .val_bits = 32, - .max_register = SUN4I_CODEC_AC_MIC_PHONE_CAL, -}; - static const struct of_device_id sun4i_codec_of_match[] = { { .compatible = "allwinner,sun4i-a10-codec" }, { .compatible = "allwinner,sun7i-a20-codec" }, @@ -586,6 +683,56 @@ static struct snd_soc_card *sun4i_codec_create_card(struct device *dev) return card; }; +/* sun7i-specific things: */ +/* MIC_PHONE_CAL register offsets and bit fields (A20 only) */ +#define SUN7I_CODEC_AC_MIC_PHONE_CAL (0x3c) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PREG1 (29) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PREG2 (26) +/* note: no idea where the output pins for the following are. */ +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PHONEOUTG (5) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PHONEOUTEN (4) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PHONEOUTS3 (3) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PHONEOUTS2 (2) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PHONEOUTS1 (1) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PHONEOUTS0 (0) + +static DECLARE_TLV_DB_RANGE(sun7i_codec_micin_preamp_gain_scale, + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 7, TLV_DB_SCALE_ITEM(2400, 300, 0) +); + +static const struct snd_kcontrol_new sun7i_codec_widgets[] = { + SUN4I_COMMON_CODEC_WIDGETS, + SOC_SINGLE_TLV("Mic1-In Boost Gain", + SUN7I_CODEC_AC_MIC_PHONE_CAL, + SUN7I_CODEC_AC_MIC_PHONE_CAL_PREG1, + 7, + 0, + sun7i_codec_micin_preamp_gain_scale), + SOC_SINGLE_TLV("Mic2-In Boost Gain", + SUN7I_CODEC_AC_MIC_PHONE_CAL, + SUN7I_CODEC_AC_MIC_PHONE_CAL_PREG2, + 7, + 0, + sun7i_codec_micin_preamp_gain_scale), +}; + +static const struct snd_soc_codec_driver sun7i_codec_codec = { + .controls = sun7i_codec_widgets, + .num_controls = ARRAY_SIZE(sun7i_codec_widgets), + .dapm_widgets = sun4i_codec_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(sun4i_codec_dapm_widgets), + .dapm_routes = sun4i_codec_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(sun4i_codec_dapm_routes), +}; +static const struct regmap_config sun4i_codec_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = SUN7I_CODEC_AC_MIC_PHONE_CAL, +}; +/* end sun7i-specific things */ + static int sun4i_codec_probe(struct platform_device *pdev) { struct snd_soc_card *card; @@ -593,6 +740,7 @@ static int sun4i_codec_probe(struct platform_device *pdev) struct resource *res; void __iomem *base; int ret; + const struct snd_soc_codec_driver* codec_codec; scodec = devm_kzalloc(&pdev->dev, sizeof(*scodec), GFP_KERNEL); if (!scodec) @@ -638,7 +786,12 @@ static int sun4i_codec_probe(struct platform_device *pdev) scodec->playback_dma_data.maxburst = 4; scodec->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; - ret = snd_soc_register_codec(&pdev->dev, &sun4i_codec_codec, + if (of_device_is_compatible(pdev->dev.of_node, + "allwinner,sun7i-a20-codec")) + codec_codec = &sun7i_codec_codec; + else + codec_codec = &sun4i_codec_codec; + ret = snd_soc_register_codec(&pdev->dev, codec_codec, &sun4i_codec_dai, 1); if (ret) { dev_err(&pdev->dev, "Failed to register our codec\n");

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