December 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v2 1/2] ASoC: mediatek: Use managed buffer allocation
by Takashi Iwai 16 Dec '19

16 Dec '19

Clean up the drivers with the new managed buffer allocation API. The superfluous snd_pcm_lib_malloc_pages() and snd_pcm_lib_free_pages() calls are dropped, as well as the superfluous snd_pcm_lib_preallocate_free_for_all() call. As of the result, hw_free and pcm_destruct ops became empty and got removed. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 6 +----- sound/soc/mediatek/common/mtk-afe-platform-driver.c | 12 ++---------- sound/soc/mediatek/common/mtk-afe-platform-driver.h | 2 -- sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 1 - sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 1 - 5 files changed, 3 insertions(+), 19 deletions(-) diff --git a/sound/soc/mediatek/common/mtk-afe-fe-dai.c b/sound/soc/mediatek/common/mtk-afe-fe-dai.c index e761cb66be5d..8f314e15ce73 100644 --- a/sound/soc/mediatek/common/mtk-afe-fe-dai.c +++ b/sound/soc/mediatek/common/mtk-afe-fe-dai.c @@ -129,10 +129,6 @@ int mtk_afe_fe_hw_params(struct snd_pcm_substream *substream, unsigned int rate = params_rate(params); snd_pcm_format_t format = params_format(params); - ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); - if (ret < 0) - return ret; - if (afe->request_dram_resource) afe->request_dram_resource(afe->dev); @@ -193,7 +189,7 @@ int mtk_afe_fe_hw_free(struct snd_pcm_substream *substream, if (afe->release_dram_resource) afe->release_dram_resource(afe->dev); - return snd_pcm_lib_free_pages(substream); + return 0; } EXPORT_SYMBOL_GPL(mtk_afe_fe_hw_free); diff --git a/sound/soc/mediatek/common/mtk-afe-platform-driver.c b/sound/soc/mediatek/common/mtk-afe-platform-driver.c index b6624d8d084b..719d43bb0b8f 100644 --- a/sound/soc/mediatek/common/mtk-afe-platform-driver.c +++ b/sound/soc/mediatek/common/mtk-afe-platform-driver.c @@ -120,25 +120,17 @@ int mtk_afe_pcm_new(struct snd_soc_component *component, struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component); size = afe->mtk_afe_hardware->buffer_bytes_max; - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - afe->dev, size, size); + snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV, + afe->dev, size, size); return 0; } EXPORT_SYMBOL_GPL(mtk_afe_pcm_new); -void mtk_afe_pcm_free(struct snd_soc_component *component, - struct snd_pcm *pcm) -{ - snd_pcm_lib_preallocate_free_for_all(pcm); -} -EXPORT_SYMBOL_GPL(mtk_afe_pcm_free); - const struct snd_soc_component_driver mtk_afe_pcm_platform = { .name = AFE_PCM_NAME, .ioctl = snd_soc_pcm_lib_ioctl, .pointer = mtk_afe_pcm_pointer, .pcm_construct = mtk_afe_pcm_new, - .pcm_destruct = mtk_afe_pcm_free, }; EXPORT_SYMBOL_GPL(mtk_afe_pcm_platform); diff --git a/sound/soc/mediatek/common/mtk-afe-platform-driver.h b/sound/soc/mediatek/common/mtk-afe-platform-driver.h index e550d11568c3..fcc923b88f12 100644 --- a/sound/soc/mediatek/common/mtk-afe-platform-driver.h +++ b/sound/soc/mediatek/common/mtk-afe-platform-driver.h @@ -21,8 +21,6 @@ snd_pcm_uframes_t mtk_afe_pcm_pointer(struct snd_soc_component *component, struct snd_pcm_substream *substream); int mtk_afe_pcm_new(struct snd_soc_component *component, struct snd_soc_pcm_runtime *rtd); -void mtk_afe_pcm_free(struct snd_soc_component *component, - struct snd_pcm *pcm); int mtk_afe_combine_sub_dai(struct mtk_base_afe *afe); int mtk_afe_add_sub_dai_control(struct snd_soc_component *component); diff --git a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c index 033c07fb599c..0743c699d8c1 100644 --- a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c +++ b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c @@ -715,7 +715,6 @@ static const struct snd_soc_component_driver mt6797_afe_component = { .ioctl = snd_soc_pcm_lib_ioctl, .pointer = mtk_afe_pcm_pointer, .pcm_construct = mtk_afe_pcm_new, - .pcm_destruct = mtk_afe_pcm_free, }; static int mt6797_dai_memif_register(struct mtk_base_afe *afe) diff --git a/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c b/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c index 76af09d8f1af..6b5f407de3d0 100644 --- a/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c +++ b/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c @@ -1053,7 +1053,6 @@ static const struct snd_soc_component_driver mt8183_afe_component = { .ioctl = snd_soc_pcm_lib_ioctl, .pointer = mtk_afe_pcm_pointer, .pcm_construct = mtk_afe_pcm_new, - .pcm_destruct = mtk_afe_pcm_free, }; static int mt8183_dai_memif_register(struct mtk_base_afe *afe) -- 2.16.4

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[alsa-devel] [PATCH] ASoC: wm8962: fix lambda value
by Shengjiu Wang 16 Dec '19

16 Dec '19

According to user manual, it is required that FLL_LAMBDA > 0 in all cases (Integer and Franctional modes). Fixes: 9a76f1ff6e29 ("ASoC: Add initial WM8962 CODEC driver") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/wm8962.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 3e5c69fbc33a..d9d59f45833f 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2788,7 +2788,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref, if (target % Fref == 0) { fll_div->theta = 0; - fll_div->lambda = 0; + fll_div->lambda = 1; } else { gcd_fll = gcd(target, fratio * Fref); @@ -2858,7 +2858,7 @@ static int wm8962_set_fll(struct snd_soc_component *component, int fll_id, int s return -EINVAL; } - if (fll_div.theta || fll_div.lambda) + if (fll_div.theta) fll1 |= WM8962_FLL_FRAC; /* Stop the FLL while we reconfigure */ -- 2.21.0

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[alsa-devel] [PATCH v2 00/15] ASoC: soc-core cleanup step7
by Kuninori Morimoto 16 Dec '19

16 Dec '19

Hi Mark These are v2 of codec_conf cleanup. ASoC has codec_conf, and we are using original dev_name, of_node for it to finding codec. But, we already have snd_soc_dai_link_component method. We can reuse it for codec_conf, too. This patches are for it. Big change from v1 is - COMP_CODEC_CNF(_name) + COMP_CODEC_CONF(_name) Kuninori Morimoto (15): ASoC: soc-core: support snd_soc_dai_link_component for codec_conf ASoC: fsl: imx-audmix: use snd_soc_dai_link_component for codec_conf ASoC: intel: kbl_da7219_max98927: use snd_soc_dai_link_component for codec_conf ASoC: intel: kbl_rt5663_max98927: use snd_soc_dai_link_component for codec_conf ASoC: intel: kbl_rt5663_rt5514_max98927: use snd_soc_dai_link_component for codec_conf ASoC: intel: skl_nau88l25_ssm4567: use snd_soc_dai_link_component for codec_conf ASoC: mediatek: mt8173-rt5650-rt5514: use snd_soc_dai_link_component for codec_conf ASoC: mediatek: mt8173-rt5650-rt5676: use snd_soc_dai_link_component for codec_conf ASoC: mediatek: mt8183-da7219-max98357: use snd_soc_dai_link_component for codec_conf ASoC: samsung: bells: use snd_soc_dai_link_component for codec_conf ASoC: samsung: lowland: use snd_soc_dai_link_component for codec_conf ASoC: samsung: neo1973_wm8753: use snd_soc_dai_link_component for codec_conf ASoC: samsung: speyside: use snd_soc_dai_link_component for codec_conf ASoC: ti: rx51: use snd_soc_dai_link_component for codec_conf ASoC: soc-core: remove legacy style of codec_conf include/sound/soc.h | 4 ++-- sound/soc/fsl/imx-audmix.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 8 ++++---- sound/soc/intel/boards/kbl_rt5663_max98927.c | 4 ++-- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 4 ++-- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 4 ++-- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 2 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 2 +- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 2 +- sound/soc/samsung/bells.c | 2 +- sound/soc/samsung/lowland.c | 2 +- sound/soc/samsung/neo1973_wm8753.c | 2 +- sound/soc/samsung/speyside.c | 2 +- sound/soc/soc-core.c | 12 +++++------- sound/soc/ti/rx51.c | 12 ++++++------ 15 files changed, 31 insertions(+), 33 deletions(-) -- 2.7.4

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[alsa-devel] WARNING in azx_rirb_get_response
by syzbot 16 Dec '19

16 Dec '19

Hello, syzbot found the following crash on: HEAD commit: 37d4e84f Merge tag 'ceph-for-5.5-rc2' of git://github.com/.. git tree: upstream console output: https://syzkaller.appspot.com/x/log.txt?x=10bbc9a6e00000 kernel config: https://syzkaller.appspot.com/x/.config?x=79f79de2a27d3e3d dashboard link: https://syzkaller.appspot.com/bug?extid=b3028ac3933f5c466389 compiler: gcc (GCC) 9.0.0 20181231 (experimental) userspace arch: i386 Unfortunately, I don't have any reproducer for this crash yet. IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+b3028ac3933f5c466389(a)syzkaller.appspotmail.com snd_hda_intel 0000:00:06.0: No response from codec, disabling MSI: last cmd=0x004a0000 ------------[ cut here ]------------ snd_hda_intel 0000:00:06.0: azx_get_response timeout, switching to single_cmd mode: last cmd=0x004a0000 WARNING: CPU: 2 PID: 8020 at sound/pci/hda/hda_controller.c:886 azx_rirb_get_response+0x5d6/0x720 sound/pci/hda/hda_controller.c:886 Kernel panic - not syncing: panic_on_warn set ... CPU: 2 PID: 8020 Comm: syz-executor.1 Not tainted 5.5.0-rc1-syzkaller #0 Hardware name: QEMU Standard PC (Q35 + ICH9, 2009), BIOS rel-1.12.0-59-gc9ba5276e321-prebuilt.qemu.org 04/01/2014 Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x197/0x210 lib/dump_stack.c:118 panic+0x2e3/0x75c kernel/panic.c:221 __warn.cold+0x2f/0x3e kernel/panic.c:582 report_bug+0x289/0x300 lib/bug.c:195 fixup_bug arch/x86/kernel/traps.c:174 [inline] fixup_bug arch/x86/kernel/traps.c:169 [inline] do_error_trap+0x11b/0x200 arch/x86/kernel/traps.c:267 do_invalid_op+0x37/0x50 arch/x86/kernel/traps.c:286 invalid_op+0x23/0x30 arch/x86/entry/entry_64.S:1027 RIP: 0010:azx_rirb_get_response+0x5d6/0x720 sound/pci/hda/hda_controller.c:886 Code: f6 00 00 00 49 8b 5d 00 e8 27 75 7a fb 4c 89 ef e8 ef 21 5a fe 44 89 f9 48 89 da 48 c7 c7 80 07 ce 88 48 89 c6 e8 d9 1e 4b fb <0f> 0b 48 b8 00 00 00 00 00 fc ff df 48 8b 4d 88 48 89 ca 48 c1 ea RSP: 0018:ffffc9000236f708 EFLAGS: 00010286 RAX: 0000000000000000 RBX: ffff888078f64240 RCX: 0000000000000000 RDX: 0000000000000000 RSI: ffffffff815e88f6 RDI: fffff5200046ded3 RBP: ffffc9000236f780 R08: ffff8880673d0140 R09: ffffed1005a46621 R10: ffffed1005a46620 R11: ffff88802d233107 R12: ffff888029ef0000 R13: ffff88802759a0b0 R14: dffffc0000000000 R15: 00000000004a0000 azx_get_response sound/pci/hda/hda_controller.c:994 [inline] azx_get_response+0x14c/0x180 sound/pci/hda/hda_controller.c:984 snd_hdac_bus_exec_verb_unlocked+0x4b5/0x950 sound/hda/hdac_bus.c:115 codec_exec_verb+0x13d/0x340 sound/pci/hda/hda_codec.c:52 snd_hdac_exec_verb+0x65/0x110 sound/hda/hdac_device.c:257 codec_read+0xa2/0x100 sound/hda/hdac_device.c:986 snd_hdac_codec_read+0x34/0x40 sound/hda/hdac_device.c:1015 snd_hda_codec_read include/sound/hda_codec.h:318 [inline] update_pcm_format.part.0+0x36/0xc0 sound/pci/hda/hda_codec.c:1057 update_pcm_format sound/pci/hda/hda_codec.c:1056 [inline] snd_hda_codec_setup_stream.part.0+0x355/0xa40 sound/pci/hda/hda_codec.c:1102 snd_hda_codec_setup_stream+0x47/0x60 sound/pci/hda/hda_codec.c:1086 capture_pcm_prepare+0x63/0x100 sound/pci/hda/hda_generic.c:5376 snd_hda_codec_prepare+0xa5/0x420 sound/pci/hda/hda_codec.c:3160 azx_pcm_prepare+0x53f/0x950 sound/pci/hda/hda_controller.c:195 snd_pcm_do_prepare+0x61/0xa0 sound/core/pcm_native.c:1729 snd_pcm_action_single+0x7a/0x130 sound/core/pcm_native.c:1111 snd_pcm_action_nonatomic+0x9c/0xb0 sound/core/pcm_native.c:1227 snd_pcm_prepare+0x121/0x1b0 sound/core/pcm_native.c:1776 snd_pcm_kernel_ioctl+0x117/0x1f0 sound/core/pcm_native.c:3068 snd_pcm_oss_prepare+0x49/0x150 sound/core/oss/pcm_oss.c:1125 snd_pcm_oss_make_ready+0x12b/0x170 sound/core/oss/pcm_oss.c:1153 snd_pcm_oss_sync.isra.0+0x1c4/0x7e0 sound/core/oss/pcm_oss.c:1707 snd_pcm_oss_release+0x214/0x290 sound/core/oss/pcm_oss.c:2545 __fput+0x2ff/0x890 fs/file_table.c:280 ____fput+0x16/0x20 fs/file_table.c:313 task_work_run+0x145/0x1c0 kernel/task_work.c:113 tracehook_notify_resume include/linux/tracehook.h:188 [inline] exit_to_usermode_loop+0x316/0x380 arch/x86/entry/common.c:164 prepare_exit_to_usermode arch/x86/entry/common.c:195 [inline] syscall_return_slowpath arch/x86/entry/common.c:278 [inline] do_syscall_32_irqs_on arch/x86/entry/common.c:352 [inline] do_fast_syscall_32+0xbbd/0xe16 arch/x86/entry/common.c:408 entry_SYSENTER_compat+0x70/0x7f arch/x86/entry/entry_64_compat.S:139 RIP: 0023:0xf7faea39 Code: 00 00 00 89 d3 5b 5e 5f 5d c3 b8 80 96 98 00 eb c4 8b 04 24 c3 8b 1c 24 c3 8b 34 24 c3 8b 3c 24 c3 51 52 55 89 e5 0f 34 cd 80 <5d> 5a 59 c3 90 90 90 90 eb 0d 90 90 90 90 90 90 90 90 90 90 90 90 RSP: 002b:00000000ffadef9c EFLAGS: 00000293 ORIG_RAX: 0000000000000006 RAX: 0000000000000000 RBX: 0000000000000004 RCX: 0000000000000000 RDX: 000000000812b000 RSI: 0000000000000005 RDI: 000000000815b660 RBP: 000000000815b660 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: 0000000000000000 R13: 0000000000000000 R14: 0000000000000000 R15: 0000000000000000 Kernel Offset: disabled Rebooting in 86400 seconds.. --- This bug is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot.

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[alsa-devel] [PATCH][resend v2] ASoC: simple-card: switch to yaml base Documentation
by Kuninori Morimoto 16 Dec '19

16 Dec '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This patch switches from .txt base to .yaml base Document. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - Add Rob to "To:" instead of "Cc:" .../devicetree/bindings/sound/simple-card.txt | 351 --------------- .../devicetree/bindings/sound/simple-card.yaml | 472 +++++++++++++++++++++ 2 files changed, 472 insertions(+), 351 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/simple-card.txt create mode 100644 Documentation/devicetree/bindings/sound/simple-card.yaml diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt deleted file mode 100644 index 79954cd6..0000000 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ /dev/null @@ -1,351 +0,0 @@ -Simple-Card: - -Simple-Card specifies audio DAI connections of SoC <-> codec. - -Required properties: - -- compatible : "simple-audio-card" - -Optional properties: - -- simple-audio-card,name : User specified audio sound card name, one string - property. -- simple-audio-card,widgets : Please refer to widgets.txt. -- simple-audio-card,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. -- simple-audio-card,mclk-fs : Multiplication factor between stream rate and codec - mclk. When defined, mclk-fs property defined in - dai-link sub nodes are ignored. -- simple-audio-card,hp-det-gpio : Reference to GPIO that signals when - headphones are attached. -- simple-audio-card,mic-det-gpio : Reference to GPIO that signals when - a microphone is attached. -- simple-audio-card,aux-devs : List of phandles pointing to auxiliary devices, such - as amplifiers, to be added to the sound card. -- simple-audio-card,pin-switches : List of strings containing the widget names for - which pin switches must be created. - -Optional subnodes: - -- simple-audio-card,dai-link : Container for dai-link level - properties and the CPU and CODEC - sub-nodes. This container may be - omitted when the card has only one - DAI link. See the examples and the - section below. - -Dai-link subnode properties and subnodes: - -If dai-link subnode is omitted and the subnode properties are directly -under "sound"-node the subnode property and subnode names have to be -prefixed with "simple-audio-card,"-prefix. - -Required dai-link subnodes: - -- cpu : CPU sub-node -- codec : CODEC sub-node - -Optional dai-link subnode properties: - -- format : CPU/CODEC common audio format. - "i2s", "right_j", "left_j" , "dsp_a" - "dsp_b", "ac97", "pdm", "msb", "lsb" -- frame-master : Indicates dai-link frame master. - phandle to a cpu or codec subnode. -- bitclock-master : Indicates dai-link bit clock master. - phandle to a cpu or codec subnode. -- bitclock-inversion : bool property. Add this if the - dai-link uses bit clock inversion. -- frame-inversion : bool property. Add this if the - dai-link uses frame clock inversion. -- mclk-fs : Multiplication factor between stream - rate and codec mclk, applied only for - the dai-link. - -For backward compatibility the frame-master and bitclock-master -properties can be used as booleans in codec subnode to indicate if the -codec is the dai-link frame or bit clock master. In this case there -should be no dai-link node, the same properties should not be present -at sound-node level, and the bitclock-inversion and frame-inversion -properties should also be placed in the codec node if needed. - -Required CPU/CODEC subnodes properties: - -- sound-dai : phandle and port of CPU/CODEC - -Optional CPU/CODEC subnodes properties: - -- dai-tdm-slot-num : Please refer to tdm-slot.txt. -- dai-tdm-slot-width : Please refer to tdm-slot.txt. -- clocks / system-clock-frequency : specify subnode's clock if needed. - it can be specified via "clocks" if system has - clock node (= common clock), or "system-clock-frequency" - (if system doens't support common clock) - If a clock is specified, it is - enabled with clk_prepare_enable() - in dai startup() and disabled with - clk_disable_unprepare() in dai - shutdown(). - If a clock is specified and a - multiplication factor is given with - mclk-fs, the clock will be set to the - calculated mclk frequency when the - stream starts. -- system-clock-direction-out : specifies clock direction as 'out' on - initialization. It is useful for some aCPUs with - fixed clocks. - -------------------------------------------- -Example 1 - single DAI link: -------------------------------------------- - -sound { - compatible = "simple-audio-card"; - simple-audio-card,name = "VF610-Tower-Sound-Card"; - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&dailink0_master>; - simple-audio-card,frame-master = <&dailink0_master>; - simple-audio-card,widgets = - "Microphone", "Microphone Jack", - "Headphone", "Headphone Jack", - "Speaker", "External Speaker"; - simple-audio-card,routing = - "MIC_IN", "Microphone Jack", - "Headphone Jack", "HP_OUT", - "External Speaker", "LINE_OUT"; - - simple-audio-card,cpu { - sound-dai = <&sh_fsi2 0>; - }; - - dailink0_master: simple-audio-card,codec { - sound-dai = <&ak4648>; - clocks = <&osc>; - }; -}; - -&i2c0 { - ak4648: ak4648@12 { - #sound-dai-cells = <0>; - compatible = "asahi-kasei,ak4648"; - reg = <0x12>; - }; -}; - -sh_fsi2: sh_fsi2@ec230000 { - #sound-dai-cells = <1>; - compatible = "renesas,sh_fsi2"; - reg = <0xec230000 0x400>; - interrupt-parent = <&gic>; - interrupts = <0 146 0x4>; -}; - -------------------------------------------- -Example 2 - many DAI links: -------------------------------------------- - -sound { - compatible = "simple-audio-card"; - simple-audio-card,name = "Cubox Audio"; - - simple-audio-card,dai-link@0 { /* I2S - HDMI */ - reg = <0>; - format = "i2s"; - cpu { - sound-dai = <&audio1 0>; - }; - codec { - sound-dai = <&tda998x 0>; - }; - }; - - simple-audio-card,dai-link@1 { /* S/PDIF - HDMI */ - reg = <1>; - cpu { - sound-dai = <&audio1 1>; - }; - codec { - sound-dai = <&tda998x 1>; - }; - }; - - simple-audio-card,dai-link@2 { /* S/PDIF - S/PDIF */ - reg = <2>; - cpu { - sound-dai = <&audio1 1>; - }; - codec { - sound-dai = <&spdif_codec>; - }; - }; -}; - -------------------------------------------- -Example 3 - route audio from IMX6 SSI2 through TLV320DAC3100 codec -through TPA6130A2 amplifier to headphones: -------------------------------------------- - -&i2c0 { - codec: tlv320dac3100@18 { - compatible = "ti,tlv320dac3100"; - ... - } - - amp: tpa6130a2@60 { - compatible = "ti,tpa6130a2"; - ... - } -} - -sound { - compatible = "simple-audio-card"; - ... - simple-audio-card,widgets = - "Headphone", "Headphone Jack"; - simple-audio-card,routing = - "Headphone Jack", "HPLEFT", - "Headphone Jack", "HPRIGHT", - "LEFTIN", "HPL", - "RIGHTIN", "HPR"; - simple-audio-card,aux-devs = <&amp>; - simple-audio-card,cpu { - sound-dai = <&ssi2>; - }; - simple-audio-card,codec { - sound-dai = <&codec>; - clocks = ... - }; -}; - -------------------------------------------- -Example 4. Sampling Rate Conversion -------------------------------------------- - -sound { - compatible = "simple-audio-card"; - - simple-audio-card,name = "rsnd-ak4643"; - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&sndcodec>; - simple-audio-card,frame-master = <&sndcodec>; - - simple-audio-card,convert-rate = <48000>; - - simple-audio-card,prefix = "ak4642"; - simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", - "DAI0 Capture", "ak4642 Capture"; - - sndcpu: simple-audio-card,cpu { - sound-dai = <&rcar_sound>; - }; - - sndcodec: simple-audio-card,codec { - sound-dai = <&ak4643>; - system-clock-frequency = <11289600>; - }; -}; - -------------------------------------------- -Example 5. 2 CPU 1 Codec (Mixing) -------------------------------------------- -sound { - compatible = "simple-audio-card"; - - simple-audio-card,name = "rsnd-ak4643"; - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&dpcmcpu>; - simple-audio-card,frame-master = <&dpcmcpu>; - - simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", - "ak4642 Playback", "DAI1 Playback"; - - dpcmcpu: cpu@0 { - sound-dai = <&rcar_sound 0>; - }; - - cpu@1 { - sound-dai = <&rcar_sound 1>; - }; - - codec { - prefix = "ak4642"; - sound-dai = <&ak4643>; - clocks = <&audio_clock>; - }; -}; - -------------------------------------------- -Example 6 - many DAI links with DPCM: -------------------------------------------- - -CPU0 ------ ak4613 -CPU1 ------ PCM3168A-p /* DPCM 1ch/2ch */ -CPU2 --/ /* DPCM 3ch/4ch */ -CPU3 --/ /* DPCM 5ch/6ch */ -CPU4 --/ /* DPCM 7ch/8ch */ -CPU5 ------ PCM3168A-c - -sound { - compatible = "simple-audio-card"; - - simple-audio-card,routing = - "pcm3168a Playback", "DAI1 Playback", - "pcm3168a Playback", "DAI2 Playback", - "pcm3168a Playback", "DAI3 Playback", - "pcm3168a Playback", "DAI4 Playback"; - - simple-audio-card,dai-link@0 { - format = "left_j"; - bitclock-master = <&sndcpu0>; - frame-master = <&sndcpu0>; - - sndcpu0: cpu { - sound-dai = <&rcar_sound 0>; - }; - codec { - sound-dai = <&ak4613>; - }; - }; - simple-audio-card,dai-link@1 { - format = "i2s"; - bitclock-master = <&sndcpu1>; - frame-master = <&sndcpu1>; - - convert-channels = <8>; /* TDM Split */ - - sndcpu1: cpu@0 { - sound-dai = <&rcar_sound 1>; - }; - cpu@1 { - sound-dai = <&rcar_sound 2>; - }; - cpu@2 { - sound-dai = <&rcar_sound 3>; - }; - cpu@3 { - sound-dai = <&rcar_sound 4>; - }; - codec { - mclk-fs = <512>; - prefix = "pcm3168a"; - dai-tdm-slot-num = <8>; - sound-dai = <&pcm3168a 0>; - }; - }; - simple-audio-card,dai-link@2 { - format = "i2s"; - bitclock-master = <&sndcpu2>; - frame-master = <&sndcpu2>; - - sndcpu2: cpu { - sound-dai = <&rcar_sound 5>; - }; - codec { - mclk-fs = <512>; - prefix = "pcm3168a"; - sound-dai = <&pcm3168a 1>; - }; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/simple-card.yaml b/Documentation/devicetree/bindings/sound/simple-card.yaml new file mode 100644 index 0000000..ac3975f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/simple-card.yaml @@ -0,0 +1,472 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/simple-card.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Simple Audio Card Driver Device Tree Bindings + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> + +definitions: + + frame-master: + description: Indicates dai-link frame master. + $ref: /schemas/types.yaml#/definitions/phandle-array + + bitclock-master: + description: Indicates dai-link bit clock master + $ref: /schemas/types.yaml#/definitions/phandle-array + + frame-inversion: + description: dai-link uses frame clock inversion + $ref: /schemas/types.yaml#/definitions/flag + + bitclock-inversion: + description: dai-link uses bit clock inversion + $ref: /schemas/types.yaml#/definitions/flag + + mclk-fs: + description: | + Multiplication factor between stream rate and codec mclk. + When defined, mclk-fs property defined in dai-link sub nodes are ignored. + $ref: /schemas/types.yaml#/definitions/uint32 + + hp-det-gpio: + description: | + Reference to GPIO that signals when headphones are attached. + $ref: /schemas/types.yaml#/definitions/phandle + + aux-devs: + description: | + List of phandles pointing to auxiliary devices, such + as amplifiers, to be added to the sound card. + $ref: /schemas/types.yaml#/definitions/phandle-array + + mic-det-gpio: + description: | + Reference to GPIO that signals when a microphone is attached. + $ref: /schemas/types.yaml#/definitions/phandle + + convert-rate: + description: CPU to Codec rate convert. + $ref: /schemas/types.yaml#/definitions/uint32 + + convert-channels: + description: CPU to Codec rate channels. + $ref: /schemas/types.yaml#/definitions/uint32 + + prefix: + description: "device name prefix" + $ref: /schemas/types.yaml#/definitions/string + + pin-switches: + description: the widget names for which pin switches must be created. + $ref: /schemas/types.yaml#/definitions/string-array + + format: + description: audio format. + oneOf: + - items: + enum: + - i2s + - right_j + - left_j + - dsp_a + - dsp_b + - ac97 + - pdm + - msb + - lsb + + dai: + type: object + properties: + sound-dai: + $ref: /schemas/types.yaml#/definitions/phandle-array + + # common properties + mclk-fs: + $ref: "#/definitions/mclk-fs" + prefix: + $ref: "#/definitions/prefix" + frame-inversion: + $ref: "#/definitions/frame-inversion" + bitclock-inversion: + $ref: "#/definitions/bitclock-inversion" + frame-master: + $ref: /schemas/types.yaml#/definitions/flag + bitclock-master: + $ref: /schemas/types.yaml#/definitions/flag + + dai-tdm-slot-num: + description: see tdm-slot.txt. + $ref: /schemas/types.yaml#/definitions/uint32 + dai-tdm-slot-width: + description: see tdm-slot.txt. + $ref: /schemas/types.yaml#/definitions/uint32 + clocks: + description: | + If a clock is specified and a multiplication factor is given with + mclk-fs, the clock will be set to the calculated mclk frequency + when the stream starts. + $ref: /schemas/types.yaml#/definitions/phandle-array + system-clock-frequency: + description: | + If a clock is specified and a multiplication factor is given with + mclk-fs, the clock will be set to the calculated mclk frequency + when the stream starts. + $ref: /schemas/types.yaml#/definitions/uint32 + system-clock-direction-out: + description: | + specifies clock direction as 'out' on initialization. + It is useful for some aCPUs with fixed clocks. + $ref: /schemas/types.yaml#/definitions/flag + required: + - sound-dai + +properties: + compatible: + contains: + enum: + - simple-audio-card + - simple-scu-audio-card + + "#address-cells": true + "#size-cells": true + + simple-audio-card,name: + description: User specified audio sound card name. + $ref: /schemas/types.yaml#/definitions/string + + simple-audio-card,widgets: + description: User specified audio sound card name. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + + simple-audio-card,routing: + description: | + 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. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + +patternProperties: + "^simple-audio-card,cpu.*": + description: CPU node + $ref: "#/definitions/dai" + "^simple-audio-card,codec.*": + description: CPU node + $ref: "#/definitions/dai" + + # common properties + "^simple-audio-card,frame-master$": + $ref: "#/definitions/frame-master" + "^simple-audio-card,bitclock-master$": + $ref: "#/definitions/bitclock-master" + "^simple-audio-card,frame-inversion$": + $ref: "#/definitions/frame-inversion" + "^simple-audio-card,bitclock-inversion$": + $ref: "#/definitions/bitclock-inversion" + "^simple-audio-card,format$": + $ref: "#/definitions/format" + "^simple-audio-card,mclk-fs$": + $ref: "#/definitions/mclk-fs" + "^simple-audio-card,hp-det-gpio$": + $ref: "#/definitions/hp-det-gpio" + "^simple-audio-card,mic-det-gpio$": + $ref: "#/definitions/mic-det-gpio" + "^simple-audio-card,aux-devs$": + $ref: "#/definitions/aux-devs" + "^simple-audio-card,convert-rate$": + $ref: "#/definitions/convert-rate" + "^simple-audio-card,convert-channels$": + $ref: "#/definitions/convert-channels" + "^simple-audio-card,prefix$": + $ref: "#/definitions/prefix" + "^simple-audio-card,pin-switches$": + $ref: "#/definitions/pin-switches" + + "^simple-audio-card,dai-link": + description: | + Container for dai-link level properties and the CPU and CODEC sub-nodes. + This container may be omitted when the card has only one DAI link. + type: object + properties: + reg: + $ref: /schemas/types.yaml#/definitions/uint32 + + # common properties + frame-master: + $ref: "#/definitions/frame-master" + bitclock-master: + $ref: "#/definitions/bitclock-master" + frame-inversion: + $ref: "#/definitions/frame-inversion" + bitclock-inversion: + $ref: "#/definitions/bitclock-inversion" + format: + $ref: "#/definitions/format" + mclk-fs: + $ref: "#/definitions/mclk-fs" + hp-det-gpio: + $ref: "#/definitions/hp-det-gpio" + mic-det-gpio: + $ref: "#/definitions/mic-det-gpio" + aux-devs: + $ref: "#/definitions/aux-devs" + convert-rate: + $ref: "#/definitions/convert-rate" + convert-channels: + $ref: "#/definitions/convert-channels" + prefix: + $ref: "#/definitions/prefix" + pin-switches: + $ref: "#/definitions/pin-switches" + + patternProperties: + "^cpu.*": + $ref: "#/definitions/dai" + "^codec.*": + $ref: "#/definitions/dai" + additionalProperties: false + +required: + - compatible + +additionalProperties: false + +examples: +#-------------------- +# single DAI link +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "VF610-Tower-Sound-Card"; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&dailink0_master>; + simple-audio-card,frame-master = <&dailink0_master>; + simple-audio-card,widgets = + "Microphone", "Microphone Jack", + "Headphone", "Headphone Jack", + "Speaker", "External Speaker"; + simple-audio-card,routing = + "MIC_IN", "Microphone Jack", + "Headphone Jack", "HP_OUT", + "External Speaker", "LINE_OUT"; + + simple-audio-card,cpu { + sound-dai = <&sh_fsi2 0>; + }; + + dailink0_master: simple-audio-card,codec { + sound-dai = <&ak4648>; + clocks = <&osc>; + }; + }; + +#-------------------- +# Multi DAI links +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "Cubox Audio"; + + #address-cells = <1>; + #size-cells = <0>; + + simple-audio-card,dai-link@0 { /* I2S - HDMI */ + reg = <0>; + format = "i2s"; + cpu { + sound-dai = <&audio0>; + }; + codec { + sound-dai = <&tda998x0>; + }; + }; + + simple-audio-card,dai-link@1 { /* S/PDIF - HDMI */ + reg = <1>; + cpu { + sound-dai = <&audio1>; + }; + codec { + sound-dai = <&tda998x1>; + }; + }; + + simple-audio-card,dai-link@2 { /* S/PDIF - S/PDIF */ + reg = <2>; + cpu { + sound-dai = <&audio2>; + }; + codec { + sound-dai = <&spdif_codec>; + }; + }; + }; + +#-------------------- +# route audio from IMX6 SSI2 through TLV320DAC3100 codec +# through TPA6130A2 amplifier to headphones: +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,widgets = + "Headphone", "Headphone Jack"; + simple-audio-card,routing = + "Headphone Jack", "HPLEFT", + "Headphone Jack", "HPRIGHT", + "LEFTIN", "HPL", + "RIGHTIN", "HPR"; + simple-audio-card,aux-devs = <&amp>; + simple-audio-card,cpu { + sound-dai = <&ssi2>; + }; + simple-audio-card,codec { + sound-dai = <&codec>; + clocks = <&clocks>; + }; + }; + +#-------------------- +# Sampling Rate Conversion +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,name = "rsnd-ak4643"; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&sndcodec>; + simple-audio-card,frame-master = <&sndcodec>; + + simple-audio-card,convert-rate = <48000>; + + simple-audio-card,prefix = "ak4642"; + simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", + "DAI0 Capture", "ak4642 Capture"; + + sndcpu: simple-audio-card,cpu { + sound-dai = <&rcar_sound>; + }; + + sndcodec: simple-audio-card,codec { + sound-dai = <&ak4643>; + system-clock-frequency = <11289600>; + }; + }; + +#-------------------- +# 2 CPU 1 Codec (Mixing) +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,name = "rsnd-ak4643"; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&dpcmcpu>; + simple-audio-card,frame-master = <&dpcmcpu>; + + simple-audio-card,convert-rate = <48000>; + simple-audio-card,convert-channels = <2>; + + simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", + "ak4642 Playback", "DAI1 Playback"; + + dpcmcpu: simple-audio-card,cpu@0 { + sound-dai = <&rcar_sound 0>; + }; + + simple-audio-card,cpu@1 { + sound-dai = <&rcar_sound 1>; + }; + + simple-audio-card,codec { + prefix = "ak4642"; + sound-dai = <&ak4643>; + clocks = <&audio_clock>; + }; + }; + +#-------------------- +# Multi DAI links with DPCM: +# +# CPU0 ------ ak4613 +# CPU1 ------ PCM3168A-p /* DPCM 1ch/2ch */ +# CPU2 --/ /* DPCM 3ch/4ch */ +# CPU3 --/ /* DPCM 5ch/6ch */ +# CPU4 --/ /* DPCM 7ch/8ch */ +# CPU5 ------ PCM3168A-c +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,routing = + "pcm3168a Playback", "DAI1 Playback", + "pcm3168a Playback", "DAI2 Playback", + "pcm3168a Playback", "DAI3 Playback", + "pcm3168a Playback", "DAI4 Playback"; + + simple-audio-card,dai-link@0 { + format = "left_j"; + bitclock-master = <&sndcpu0>; + frame-master = <&sndcpu0>; + + sndcpu0: cpu { + sound-dai = <&rcar_sound 0>; + }; + codec { + sound-dai = <&ak4613>; + }; + }; + + simple-audio-card,dai-link@1 { + format = "i2s"; + bitclock-master = <&sndcpu1>; + frame-master = <&sndcpu1>; + + convert-channels = <8>; /* TDM Split */ + + sndcpu1: cpu@0 { + sound-dai = <&rcar_sound 1>; + }; + cpu@1 { + sound-dai = <&rcar_sound 2>; + }; + cpu@2 { + sound-dai = <&rcar_sound 3>; + }; + cpu@3 { + sound-dai = <&rcar_sound 4>; + }; + codec { + mclk-fs = <512>; + prefix = "pcm3168a"; + dai-tdm-slot-num = <8>; + sound-dai = <&pcm3168a 0>; + }; + }; + + simple-audio-card,dai-link@2 { + format = "i2s"; + bitclock-master = <&sndcpu2>; + frame-master = <&sndcpu2>; + + sndcpu2: cpu { + sound-dai = <&rcar_sound 5>; + }; + codec { + mclk-fs = <512>; + prefix = "pcm3168a"; + sound-dai = <&pcm3168a 1>; + }; + }; + }; -- 2.7.4

1 0

[alsa-devel] [PATCH][resend v2] ASoC: renesas, rsnd: switch to yaml base Documentation
by Kuninori Morimoto 16 Dec '19

16 Dec '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This patch switches from .txt base to .yaml base Document. It is still keeping detail explanations at .txt Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - Add Rob to "To:" instead of "Cc:" .../devicetree/bindings/sound/renesas,rsnd.txt | 518 -------------------- .../devicetree/bindings/sound/renesas,rsnd.yaml | 527 +++++++++++++++++++++ 2 files changed, 527 insertions(+), 518 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/renesas,rsnd.yaml diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 797fd03..b731f16 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -253,521 +253,3 @@ This is example of TDM 6ch. Driver can automatically switches TDM <-> stereo mode in this case. see "Example: simple sound card for TDM" - -============================================= -Required properties: -============================================= - -- compatible : "renesas,rcar_sound-<soctype>", fallbacks - "renesas,rcar_sound-gen1" if generation1, and - "renesas,rcar_sound-gen2" if generation2 (or RZ/G1) - "renesas,rcar_sound-gen3" if generation3 (or RZ/G2) - Examples with soctypes are: - - "renesas,rcar_sound-r8a7743" (RZ/G1M) - - "renesas,rcar_sound-r8a7744" (RZ/G1N) - - "renesas,rcar_sound-r8a7745" (RZ/G1E) - - "renesas,rcar_sound-r8a77470" (RZ/G1C) - - "renesas,rcar_sound-r8a774a1" (RZ/G2M) - - "renesas,rcar_sound-r8a774b1" (RZ/G2N) - - "renesas,rcar_sound-r8a774c0" (RZ/G2E) - - "renesas,rcar_sound-r8a7778" (R-Car M1A) - - "renesas,rcar_sound-r8a7779" (R-Car H1) - - "renesas,rcar_sound-r8a7790" (R-Car H2) - - "renesas,rcar_sound-r8a7791" (R-Car M2-W) - - "renesas,rcar_sound-r8a7793" (R-Car M2-N) - - "renesas,rcar_sound-r8a7794" (R-Car E2) - - "renesas,rcar_sound-r8a7795" (R-Car H3) - - "renesas,rcar_sound-r8a7796" (R-Car M3-W) - - "renesas,rcar_sound-r8a77965" (R-Car M3-N) - - "renesas,rcar_sound-r8a77990" (R-Car E3) - - "renesas,rcar_sound-r8a77995" (R-Car D3) -- reg : Should contain the register physical address. - required register is - SRU/ADG/SSI if generation1 - SRU/ADG/SSIU/SSI/AUDIO-DMAC-periperi if generation2/generation3 - Select extended AUDIO-DMAC-periperi address if SoC has it, - otherwise select normal AUDIO-DMAC-periperi address. -- reg-names : Should contain the register names. - scu/adg/ssi if generation1 - scu/adg/ssiu/ssi/audmapp if generation2/generation3 -- rcar_sound,ssi : Should contain SSI feature. - The number of SSI subnode should be same as HW. - see below for detail. -- rcar_sound,ssiu : Should contain SSIU feature. - The number of SSIU subnode should be same as HW. - see below for detail. -- rcar_sound,src : Should contain SRC feature. - The number of SRC subnode should be same as HW. - see below for detail. -- rcar_sound,ctu : Should contain CTU feature. - The number of CTU subnode should be same as HW. - see below for detail. -- rcar_sound,mix : Should contain MIX feature. - The number of MIX subnode should be same as HW. - see below for detail. -- rcar_sound,dvc : Should contain DVC feature. - The number of DVC subnode should be same as HW. - see below for detail. -- rcar_sound,dai : DAI contents. - The number of DAI subnode should be same as HW. - see below for detail. -- #sound-dai-cells : it must be 0 if your system is using single DAI - it must be 1 if your system is using multi DAI -- clocks : References to SSI/SRC/MIX/CTU/DVC/AUDIO_CLK clocks. -- clock-names : List of necessary clock names. - "ssi-all", "ssi.X", "src.X", "mix.X", "ctu.X", - "dvc.X", "clk_a", "clk_b", "clk_c", "clk_i" - -Optional properties: -- #clock-cells : it must be 0 if your system has audio_clkout - it must be 1 if your system has audio_clkout0/1/2/3 -- clock-frequency : for all audio_clkout0/1/2/3 -- clkout-lr-asynchronous : boolean property. it indicates that audio_clkoutn - is asynchronizes with lr-clock. -- resets : References to SSI resets. -- reset-names : List of valid reset names. - "ssi-all", "ssi.X" - -SSI subnode properties: -- interrupts : Should contain SSI interrupt for PIO transfer -- shared-pin : if shared clock pin -- pio-transfer : use PIO transfer mode -- no-busif : BUSIF is not ussed when [mem -> SSI] via DMA case -- dma : Should contain Audio DMAC entry -- dma-names : SSI case "rx" (=playback), "tx" (=capture) - Deprecated: see SSIU subnode properties - SSIU case "rxu" (=playback), "txu" (=capture) - -SSIU subnode properties: -- dma : Should contain Audio DMAC entry -- dma-names : "rx" (=playback), "tx" (=capture) - -SRC subnode properties: -- dma : Should contain Audio DMAC entry -- dma-names : "rx" (=playback), "tx" (=capture) - -DVC subnode properties: -- dma : Should contain Audio DMAC entry -- dma-names : "tx" (=playback/capture) - -DAI subnode properties: -- playback : list of playback modules -- capture : list of capture modules - - -============================================= -Example: -============================================= - -rcar_sound: sound@ec500000 { - #sound-dai-cells = <1>; - compatible = "renesas,rcar_sound-r8a7791", "renesas,rcar_sound-gen2"; - reg = <0 0xec500000 0 0x1000>, /* SCU */ - <0 0xec5a0000 0 0x100>, /* ADG */ - <0 0xec540000 0 0x1000>, /* SSIU */ - <0 0xec541000 0 0x1280>, /* SSI */ - <0 0xec740000 0 0x200>; /* Audio DMAC peri peri*/ - reg-names = "scu", "adg", "ssiu", "ssi", "audmapp"; - - clocks = <&mstp10_clks R8A7790_CLK_SSI_ALL>, - <&mstp10_clks R8A7790_CLK_SSI9>, <&mstp10_clks R8A7790_CLK_SSI8>, - <&mstp10_clks R8A7790_CLK_SSI7>, <&mstp10_clks R8A7790_CLK_SSI6>, - <&mstp10_clks R8A7790_CLK_SSI5>, <&mstp10_clks R8A7790_CLK_SSI4>, - <&mstp10_clks R8A7790_CLK_SSI3>, <&mstp10_clks R8A7790_CLK_SSI2>, - <&mstp10_clks R8A7790_CLK_SSI1>, <&mstp10_clks R8A7790_CLK_SSI0>, - <&mstp10_clks R8A7790_CLK_SCU_SRC9>, <&mstp10_clks R8A7790_CLK_SCU_SRC8>, - <&mstp10_clks R8A7790_CLK_SCU_SRC7>, <&mstp10_clks R8A7790_CLK_SCU_SRC6>, - <&mstp10_clks R8A7790_CLK_SCU_SRC5>, <&mstp10_clks R8A7790_CLK_SCU_SRC4>, - <&mstp10_clks R8A7790_CLK_SCU_SRC3>, <&mstp10_clks R8A7790_CLK_SCU_SRC2>, - <&mstp10_clks R8A7790_CLK_SCU_SRC1>, <&mstp10_clks R8A7790_CLK_SCU_SRC0>, - <&mstp10_clks R8A7790_CLK_SCU_DVC0>, <&mstp10_clks R8A7790_CLK_SCU_DVC1>, - <&audio_clk_a>, <&audio_clk_b>, <&audio_clk_c>, <&m2_clk>; - clock-names = "ssi-all", - "ssi.9", "ssi.8", "ssi.7", "ssi.6", "ssi.5", - "ssi.4", "ssi.3", "ssi.2", "ssi.1", "ssi.0", - "src.9", "src.8", "src.7", "src.6", "src.5", - "src.4", "src.3", "src.2", "src.1", "src.0", - "dvc.0", "dvc.1", - "clk_a", "clk_b", "clk_c", "clk_i"; - - rcar_sound,dvc { - dvc0: dvc-0 { - dmas = <&audma0 0xbc>; - dma-names = "tx"; - }; - dvc1: dvc-1 { - dmas = <&audma0 0xbe>; - dma-names = "tx"; - }; - }; - - rcar_sound,mix { - mix0: mix-0 { }; - mix1: mix-1 { }; - }; - - rcar_sound,ctu { - ctu00: ctu-0 { }; - ctu01: ctu-1 { }; - ctu02: ctu-2 { }; - ctu03: ctu-3 { }; - ctu10: ctu-4 { }; - ctu11: ctu-5 { }; - ctu12: ctu-6 { }; - ctu13: ctu-7 { }; - }; - - rcar_sound,src { - src0: src-0 { - interrupts = <0 352 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x85>, <&audma1 0x9a>; - dma-names = "rx", "tx"; - }; - src1: src-1 { - interrupts = <0 353 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x87>, <&audma1 0x9c>; - dma-names = "rx", "tx"; - }; - src2: src-2 { - interrupts = <0 354 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x89>, <&audma1 0x9e>; - dma-names = "rx", "tx"; - }; - src3: src-3 { - interrupts = <0 355 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x8b>, <&audma1 0xa0>; - dma-names = "rx", "tx"; - }; - src4: src-4 { - interrupts = <0 356 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x8d>, <&audma1 0xb0>; - dma-names = "rx", "tx"; - }; - src5: src-5 { - interrupts = <0 357 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x8f>, <&audma1 0xb2>; - dma-names = "rx", "tx"; - }; - src6: src-6 { - interrupts = <0 358 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x91>, <&audma1 0xb4>; - dma-names = "rx", "tx"; - }; - src7: src-7 { - interrupts = <0 359 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x93>, <&audma1 0xb6>; - dma-names = "rx", "tx"; - }; - src8: src-8 { - interrupts = <0 360 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x95>, <&audma1 0xb8>; - dma-names = "rx", "tx"; - }; - src9: src-9 { - interrupts = <0 361 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x97>, <&audma1 0xba>; - dma-names = "rx", "tx"; - }; - }; - - rcar_sound,ssiu { - ssiu00: ssiu-0 { - dmas = <&audma0 0x15>, <&audma1 0x16>; - dma-names = "rx", "tx"; - }; - ssiu01: ssiu-1 { - dmas = <&audma0 0x35>, <&audma1 0x36>; - dma-names = "rx", "tx"; - }; - - ... - - ssiu95: ssiu-49 { - dmas = <&audma0 0xA5>, <&audma1 0xA6>; - dma-names = "rx", "tx"; - }; - ssiu96: ssiu-50 { - dmas = <&audma0 0xA7>, <&audma1 0xA8>; - dma-names = "rx", "tx"; - }; - ssiu97: ssiu-51 { - dmas = <&audma0 0xA9>, <&audma1 0xAA>; - dma-names = "rx", "tx"; - }; - }; - - rcar_sound,ssi { - ssi0: ssi-0 { - interrupts = <0 370 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x01>, <&audma1 0x02>; - dma-names = "rx", "tx"; - }; - ssi1: ssi-1 { - interrupts = <0 371 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x03>, <&audma1 0x04>; - dma-names = "rx", "tx"; - }; - - ... - - ssi8: ssi-8 { - interrupts = <0 378 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x11>, <&audma1 0x12>; - dma-names = "rx", "tx"; - }; - ssi9: ssi-9 { - interrupts = <0 379 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x13>, <&audma1 0x14>; - dma-names = "rx", "tx"; - }; - }; - - rcar_sound,dai { - dai0 { - playback = <&ssi5 &src5>; - capture = <&ssi6>; - }; - dai1 { - playback = <&ssi3>; - }; - dai2 { - capture = <&ssi4>; - }; - dai3 { - playback = <&ssi7>; - }; - dai4 { - capture = <&ssi8>; - }; - }; -}; - -============================================= -Example: simple sound card -============================================= - - rsnd_ak4643: sound { - compatible = "simple-audio-card"; - - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&sndcodec>; - simple-audio-card,frame-master = <&sndcodec>; - - sndcpu: simple-audio-card,cpu { - sound-dai = <&rcar_sound>; - }; - - sndcodec: simple-audio-card,codec { - sound-dai = <&ak4643>; - clocks = <&audio_clock>; - }; - }; - -&rcar_sound { - pinctrl-0 = <&sound_pins &sound_clk_pins>; - pinctrl-names = "default"; - - /* Single DAI */ - #sound-dai-cells = <0>; - - - rcar_sound,dai { - dai0 { - playback = <&ssi0 &src2 &dvc0>; - capture = <&ssi1 &src3 &dvc1>; - }; - }; -}; - -&ssi1 { - shared-pin; -}; - -============================================= -Example: simple sound card for Asynchronous mode -============================================= - -sound { - compatible = "simple-scu-audio-card"; - ... - /* - * SRC Asynchronous mode setting - * Playback: - * All input data will be converted to 48kHz - * Capture: - * Inputed 48kHz data will be converted to - * system specified Hz - */ - simple-audio-card,convert-rate = <48000>; - ... - simple-audio-card,cpu { - sound-dai = <&rcar_sound>; - }; - simple-audio-card,codec { - ... - }; -}; - -============================================= -Example: simple sound card for channel convert -============================================= - -sound { - compatible = "simple-scu-audio-card"; - ... - /* - * CTU setting - * All input data will be converted to 2ch - * as output data - */ - simple-audio-card,convert-channels = <2>; - ... - simple-audio-card,cpu { - sound-dai = <&rcar_sound>; - }; - simple-audio-card,codec { - ... - }; -}; - -============================================= -Example: simple sound card for MIXer -============================================= - -sound { - compatible = "simple-scu-audio-card"; - ... - simple-audio-card,cpu@0 { - sound-dai = <&rcar_sound 0>; - }; - simple-audio-card,cpu@1 { - sound-dai = <&rcar_sound 1>; - }; - simple-audio-card,codec { - ... - }; -}; - -&rcar_sound { - ... - rcar_sound,dai { - dai0 { - playback = <&src1 &ctu02 &mix0 &dvc0 &ssi0>; - }; - dai1 { - playback = <&src2 &ctu03 &mix0 &dvc0 &ssi0>; - }; - }; -}; - -============================================= -Example: simple sound card for TDM -============================================= - -rsnd_tdm: sound { - compatible = "simple-audio-card"; - - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&sndcodec>; - simple-audio-card,frame-master = <&sndcodec>; - - sndcpu: simple-audio-card,cpu { - sound-dai = <&rcar_sound>; - dai-tdm-slot-num = <6>; - }; - - sndcodec: simple-audio-card,codec { - sound-dai = <&xxx>; - }; -}; - -============================================= -Example: simple sound card for TDM Split -============================================= - -sound_card: sound { - compatible = "audio-graph-scu-card"; - prefix = "xxxx"; - routing = "xxxx Playback", "DAI0 Playback", - "xxxx Playback", "DAI1 Playback", - "xxxx Playback", "DAI2 Playback", - "xxxx Playback", "DAI3 Playback"; - convert-channels = <8>; /* TDM Split */ - - dais = <&rsnd_port0 /* playback ch1/ch2 */ - &rsnd_port1 /* playback ch3/ch4 */ - &rsnd_port2 /* playback ch5/ch6 */ - &rsnd_port3 /* playback ch7/ch8 */ - >; -}; - -audio-codec { - ... - port { - codec_0: endpoint@1 { - remote-endpoint = <&rsnd_ep0>; - }; - codec_1: endpoint@2 { - remote-endpoint = <&rsnd_ep1>; - }; - codec_2: endpoint@3 { - remote-endpoint = <&rsnd_ep2>; - }; - codec_3: endpoint@4 { - remote-endpoint = <&rsnd_ep3>; - }; - }; -}; - -&rcar_sound { - ... - ports { - rsnd_port0: port@0 { - rsnd_ep0: endpoint { - remote-endpoint = <&codec_0>; - ... - playback = <&ssiu30 &ssi3>; - }; - }; - rsnd_port1: port@1 { - rsnd_ep1: endpoint { - remote-endpoint = <&codec_1>; - ... - playback = <&ssiu31 &ssi3>; - }; - }; - rsnd_port2: port@2 { - rsnd_ep2: endpoint { - remote-endpoint = <&codec_2>; - ... - playback = <&ssiu32 &ssi3>; - }; - }; - rsnd_port3: port@3 { - rsnd_ep3: endpoint { - remote-endpoint = <&codec_3>; - ... - playback = <&ssiu33 &ssi3>; - }; - }; - }; -}; - -============================================= -Example: simple sound card for Multi channel -============================================= - -&rcar_sound { - pinctrl-0 = <&sound_pins &sound_clk_pins>; - pinctrl-names = "default"; - - /* Single DAI */ - #sound-dai-cells = <0>; - - - rcar_sound,dai { - dai0 { - playback = <&ssi0 &ssi1 &ssi2 &src0 &dvc0>; - }; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml new file mode 100644 index 0000000..5fb036c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -0,0 +1,527 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/renesas,rsnd.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Renesas R-Car Sound Driver Device Tree Bindings + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> + +properties: + $nodename: + pattern: "^sound@.*" + + compatible: + oneOf: + # for Gen1 SoC + - items: + - enum: + - renesas,rcar_sound-r8a7778 # R-Car M1A + - renesas,rcar_sound-r8a7779 # R-Car H1 + - enum: + - renesas,rcar_sound-gen1 + # for Gen2 SoC + - items: + - enum: + - renesas,rcar_sound-r8a7743 # RZ/G1M + - renesas,rcar_sound-r8a7744 # RZ/G1N + - renesas,rcar_sound-r8a7745 # RZ/G1E + - renesas,rcar_sound-r8a77470 # RZ/G1C + - renesas,rcar_sound-r8a7790 # R-Car H2 + - renesas,rcar_sound-r8a7791 # R-Car M2-W + - renesas,rcar_sound-r8a7793 # R-Car M2-N + - renesas,rcar_sound-r8a7794 # R-Car E2 + - enum: + - renesas,rcar_sound-gen2 + # for Gen3 SoC + - items: + - enum: + - renesas,rcar_sound-r8a774a1 # RZ/G2M + - renesas,rcar_sound-r8a774b1 # RZ/G2N + - renesas,rcar_sound-r8a774c0 # RZ/G2E + - renesas,rcar_sound-r8a7795 # R-Car H3 + - renesas,rcar_sound-r8a7796 # R-Car M3-W + - renesas,rcar_sound-r8a77965 # R-Car M3-N + - renesas,rcar_sound-r8a77990 # R-Car E3 + - renesas,rcar_sound-r8a77995 # R-Car D3 + - enum: + - renesas,rcar_sound-gen3 + # for Generic + - items: + - enum: + - renesas,rcar_sound-gen1 + - renesas,rcar_sound-gen2 + - renesas,rcar_sound-gen3 + + reg: + minItems: 3 # Gen1 + maxItems: 5 # Gen2/Gen3 + + reg-names: + description: | + Should contain the register names. + scu/adg/ssi if generation1 + scu/adg/ssiu/ssi/audmapp if generation2/generation3 + minItems: 3 # Gen1 + maxItems: 5 # Gen2/Gen3 + allOf: + - items: + enum: + - scu + - adg + - ssiu + - ssi + - audmapp + + "#sound-dai-cells": + description: | + it must be 0 if your system is using single DAI + it must be 1 if your system is using multi DAIs + enum: [0, 1] + + "#clock-cells": + description: | + it must be 0 if your system has audio_clkout + it must be 1 if your system has audio_clkout0/1/2/3 + enum: [0, 1] + + clock-frequency: + description: for audio_clkout0/1/2/3 + $ref: /schemas/types.yaml#/definitions/uint32-array + + clkout-lr-asynchronous: + description: audio_clkoutn is asynchronizes with lr-clock. + $ref: /schemas/types.yaml#/definitions/flag + + power-domains: true + + resets: + description: References to SSI resets + $ref: /schemas/types.yaml#/definitions/phandle-array + + reset-names: + description: List of valid reset names. + allOf: + - items: + enum: + - ssi-all + - ssi.9 + - ssi.8 + - ssi.7 + - ssi.6 + - ssi.5 + - ssi.4 + - ssi.3 + - ssi.2 + - ssi.1 + - ssi.0 + + clocks: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: References to SSI/SRC/MIX/CTU/DVC/AUDIO_CLK clocks. + + clock-names: + description: List of necessary clock names. + allOf: + - items: + enum: + - ssi-all + - ssi.9 + - ssi.8 + - ssi.7 + - ssi.6 + - ssi.5 + - ssi.4 + - ssi.3 + - ssi.2 + - ssi.1 + - ssi.0 + - src.9 + - src.8 + - src.7 + - src.6 + - src.5 + - src.4 + - src.3 + - src.2 + - src.1 + - src.0 + - mix.1 + - mix.0 + - ctu.1 + - ctu.0 + - dvc.1 + - dvc.0 + - clk_a + - clk_b + - clk_c + - clk_i + + # For OF-graph + port: + description: OF-Graph subnode + type: object + properties: + reg: + $ref: /schemas/types.yaml#/definitions/uint32 + endpoint: + type: object + properties: + remote-endpoint: + $ref: /schemas/types.yaml#/definitions/phandle-array + dai-format: + $ref: "simple-card.yaml#/definitions/format" + playback: + $ref: /schemas/types.yaml#/definitions/phandle-array + capture: + $ref: /schemas/types.yaml#/definitions/phandle-array + required: + - remote-endpoint + + # For multi OF-graph + ports: + description: multi OF-Graph subnode + type: object + patternProperties: + "port.*": + $ref: "#properties/port" + +patternProperties: + "^rcar_sound,dvc$": + description: DVC subnode. + type: object + patternProperties: + "dvc.*": + type: object + properties: + dmas: + $ref: /schemas/types.yaml#/definitions/phandle-array + dma-names: + const: "tx" + required: + - dmas + - dma-names + additionalProperties: false + + "^rcar_sound,mix$": + description: MIX subnode. + type: object + patternProperties: + "mix.*": + type: object + additionalProperties: false + + "^rcar_sound,ctu$": + description: CTU subnode. + type: object + patternProperties: + "ctu.*": + type: object + additionalProperties: false + + "^rcar_sound,src$": + description: SRC subnode. + type: object + patternProperties: + "src.*": + type: object + properties: + interrupts: + maxItems: 1 + dmas: + $ref: /schemas/types.yaml#/definitions/phandle-array + dma-names: + allOf: + - items: + enum: + - tx + - rx + if: + properties: + status: + const: "ok" + then: + required: + - interrupts + - dmas + - dma-names + additionalProperties: false + + "^rcar_sound,ssiu$": + description: SSIU subnode. + type: object + patternProperties: + "ssiu.*": + type: object + properties: + dmas: + $ref: /schemas/types.yaml#/definitions/phandle-array + dma-names: + allOf: + - items: + enum: + - tx + - rx + required: + - dmas + - dma-names + additionalProperties: false + + "^rcar_sound,ssi$": + description: SSI subnode. + type: object + patternProperties: + "ssi.*": + type: object + properties: + interrupts: + maxItems: 1 + dmas: + $ref: /schemas/types.yaml#/definitions/phandle-array + dma-names: + allOf: + - items: + enum: + - tx + - rx + - txu # if no ssiu node + - rxu # if no ssiu node + + shared-pin: + description: shared clock pin + $ref: /schemas/types.yaml#/definitions/flag + pio-transfer: + description: PIO transfer mode + $ref: /schemas/types.yaml#/definitions/flag + no-busif: + description: BUSIF is not used when [mem -> SSI] via DMA case + $ref: /schemas/types.yaml#/definitions/flag + required: + - interrupts + - dmas + - dma-names + additionalProperties: false + + # For DAI base + "^rcar_sound,dai$": + description: DAI subnode. + type: object + patternProperties: + "dai.*": + type: object + properties: + playback: + $ref: /schemas/types.yaml#/definitions/phandle-array + capture: + $ref: /schemas/types.yaml#/definitions/phandle-array + anyOf: + - required: + - playback + - required: + - capture + additionalProperties: false + +required: + - compatible + - reg + - reg-names + - clocks + - clock-names + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + rcar_sound: sound@ec500000 { + #sound-dai-cells = <1>; + compatible = "renesas,rcar_sound-r8a7790", "renesas,rcar_sound-gen2"; + reg = <0 0xec500000 0 0x1000>, /* SCU */ + <0 0xec5a0000 0 0x100>, /* ADG */ + <0 0xec540000 0 0x1000>, /* SSIU */ + <0 0xec541000 0 0x1280>, /* SSI */ + <0 0xec740000 0 0x200>; /* Audio DMAC peri peri*/ + reg-names = "scu", "adg", "ssiu", "ssi", "audmapp"; + + clocks = <&mstp10_clks 1005>, /* SSI-ALL */ + <&mstp10_clks 1006>, <&mstp10_clks 1007>, /* SSI9, SSI8 */ + <&mstp10_clks 1008>, <&mstp10_clks 1009>, /* SSI7, SSI6 */ + <&mstp10_clks 1010>, <&mstp10_clks 1011>, /* SSI5, SSI4 */ + <&mstp10_clks 1012>, <&mstp10_clks 1013>, /* SSI3, SSI2 */ + <&mstp10_clks 1014>, <&mstp10_clks 1015>, /* SSI1, SSI0 */ + <&mstp10_clks 1022>, <&mstp10_clks 1023>, /* SRC9, SRC8 */ + <&mstp10_clks 1024>, <&mstp10_clks 1025>, /* SRC7, SRC6 */ + <&mstp10_clks 1026>, <&mstp10_clks 1027>, /* SRC5, SRC4 */ + <&mstp10_clks 1028>, <&mstp10_clks 1029>, /* SRC3, SRC2 */ + <&mstp10_clks 1030>, <&mstp10_clks 1031>, /* SRC1, SRC0 */ + <&mstp10_clks 1020>, <&mstp10_clks 1021>, /* DVC0, DVC1 */ + <&mstp10_clks 1020>, <&mstp10_clks 1021>, /* CLKA, CLKB */ + <&mstp10_clks 1019>, <&mstp10_clks 1018>; /* CLKC, CLKI */ + + clock-names = "ssi-all", + "ssi.9", "ssi.8", + "ssi.7", "ssi.6", + "ssi.5", "ssi.4", + "ssi.3", "ssi.2", + "ssi.1", "ssi.0", + "src.9", "src.8", + "src.7", "src.6", + "src.5", "src.4", + "src.3", "src.2", + "src.1", "src.0", + "dvc.0", "dvc.1", + "clk_a", "clk_b", + "clk_c", "clk_i"; + + rcar_sound,dvc { + dvc0: dvc-0 { + dmas = <&audma0 0xbc>; + dma-names = "tx"; + }; + dvc1: dvc-1 { + dmas = <&audma0 0xbe>; + dma-names = "tx"; + }; + }; + + + rcar_sound,mix { + mix0: mix-0 { }; + mix1: mix-1 { }; + }; + + rcar_sound,ctu { + ctu00: ctu-0 { }; + ctu01: ctu-1 { }; + ctu02: ctu-2 { }; + ctu03: ctu-3 { }; + ctu10: ctu-4 { }; + ctu11: ctu-5 { }; + ctu12: ctu-6 { }; + ctu13: ctu-7 { }; + }; + + rcar_sound,src { + src0: src-0 { + status = "disabled"; + }; + src1: src-1 { + interrupts = <0 353 0>; + dmas = <&audma0 0x87>, <&audma1 0x9c>; + dma-names = "rx", "tx"; + }; + /* skip after src-2 */ + }; + + rcar_sound,ssiu { + ssiu00: ssiu-0 { + dmas = <&audma0 0x15>, <&audma1 0x16>; + dma-names = "rx", "tx"; + }; + ssiu01: ssiu-1 { + dmas = <&audma0 0x35>, <&audma1 0x36>; + dma-names = "rx", "tx"; + }; + /* skip after ssiu-2 */ + }; + + rcar_sound,ssi { + ssi0: ssi-0 { + interrupts = <0 370 1>; + dmas = <&audma0 0x01>, <&audma1 0x02>; + dma-names = "rx", "tx"; + }; + ssi1: ssi-1 { + interrupts = <0 371 1>; + dmas = <&audma0 0x03>, <&audma1 0x04>; + dma-names = "rx", "tx"; + }; + /* skip other ssi-2 */ + }; + + /* DAI base */ + rcar_sound,dai { + dai0 { + playback = <&ssi5 &src5>; + capture = <&ssi6>; + }; + dai1 { + playback = <&ssi3>; + }; + dai2 { + capture = <&ssi4>; + }; + dai3 { + playback = <&ssi7>; + }; + dai4 { + capture = <&ssi8>; + }; + }; + + /* OF-Graph */ + port { + rsnd_endpoint: endpoint { + remote-endpoint = <&codec_endpoint>; + + dai-format = "left_j"; + bitclock-master = <&rsnd_endpoint0>; + frame-master = <&rsnd_endpoint0>; + + playback = <&ssi0 &src0 &dvc0>; + capture = <&ssi1 &src1 &dvc1>; + }; + }; + + /* Multi OF-Graph */ + ports { + #address-cells = <1>; + #size-cells = <0>; + rsnd_port0: port@0 { + reg = <0>; + rsnd_endpoint0: endpoint { + remote-endpoint = <&codec_endpoint1>; + + dai-format = "left_j"; + bitclock-master = <&rsnd_endpoint0>; + frame-master = <&rsnd_endpoint0>; + + playback = <&ssi0 &src0 &dvc0>; + capture = <&ssi1 &src1 &dvc1>; + }; + }; + rsnd_port1: port@1 { + reg = <1>; + rsnd_endpoint1: endpoint { + remote-endpoint = <&codec_endpoint2>; + + dai-format = "i2s"; + playback = <&ssi0 &src0 &dvc0>; + capture = <&ssi1 &src1 &dvc1>; + }; + }; + }; + }; + + + codec { + ports { + port@0 { + codec_endpoint: endpoint { + remote-endpoint = <&rsnd_endpoint>; + }; + }; + port@1 { + codec_endpoint1: endpoint { + remote-endpoint = <&rsnd_endpoint0>; + }; + }; + port@2 { + codec_endpoint2: endpoint { + remote-endpoint = <&rsnd_endpoint1>; + }; + }; + }; + }; -- 2.7.4

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[alsa-devel] Don't use 16 bit signed integers for sample rate conversion.
by GitHub issues - edited 16 Dec '19

16 Dec '19

alsa-project/alsa-plugins issue #5 was edited from crocket: Sample rate conversion should be done with the highest resolution possible. Maybe, double, unsigned int, or unsigned int64? Once sample rate conversion is done, using S32_LE is meaningless. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/5 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[alsa-devel] Don't use 16 bit signed integers for sample rate conversion.
by GitHub issues - edited 16 Dec '19

16 Dec '19

alsa-project/alsa-plugins issue #5 was edited from crocket: Sample rate conversion should be done with the highest resolution possible. Maybe, double? Once sample rate conversion is done, using S32_LE is meaningless. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/5 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[alsa-devel] Don't use 16 bit signed integers for sample rate conversion.
by GitHub issues - edited 16 Dec '19

16 Dec '19

alsa-project/alsa-plugins issue #5 was edited from crocket: Sample rate conversion should be done with the highest resolution possible. Once sample rate conversion is done, using S32_LE is meaningless. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/5 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[alsa-devel] Don't use 16 bits signed integers for sample rate conversion.
by GitHub issues - opened 16 Dec '19

16 Dec '19

alsa-project/alsa-plugins issue #5 was opened from crocket: Sample rate conversion should be done with the highest resolution possible. Once sample rate conversion is done, using S32_LE is meaningless. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/5 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0