November 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [RFC][PATCH] ASoC: rsnd: add SSIU BUSIF support
by Kuninori Morimoto 15 Nov '18

15 Nov '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Gen2 has BUSIF0-3, Gen3 has BUSIF0-7 on some SSIU. Current driver is assuming it is using BUSIF0 as default. Thus, SSI is attaching SSIU (with BUSIF0) by using rsnd_ssiu_attach(). But, TDM split mode also needs other BUSIF to use it. This patch adds missing SSIU BUSIFx support. BUSIF is handled by SSIU instead of SSI anymore. Thus, its settings no longer needed on SSI node on DT. This patch removes its settings from Document, but driver is still keeping compatibility. Thus, old DT style is still working. But, to avoid confusing, it doesn't indicate old compatiblity things on Document. New SoC should have SSIU on DT from this patch. 1) old style DT is still supported (= no rcar_sound,ssiu node on DT) 2) If ssiu is not indicated on playback/capture, BUSIF0 will be used as default playback = <&ssi3>; /* ssiu30 will be selected */ 3) you can select own ssiu playback = <&ssi32 &ssi3>; /* ssiu32 will be selected */ Not-yet-Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- > Jiada This patch supports BUSIF, you can select it on DT. Is this style OK for you ? For example playback = <&src0 &dvc0 &ssiu02 &ssi0>; ~~~~~~~ It is keeping compatiblity. no <&ssiu> is still OK playback = <&src0 &dvc0 &ssi0>; .../devicetree/bindings/sound/renesas,rsnd.txt | 78 ++++----- sound/soc/sh/rcar/core.c | 1 + sound/soc/sh/rcar/dma.c | 37 ++++- sound/soc/sh/rcar/rsnd.h | 6 +- sound/soc/sh/rcar/ssi.c | 22 +-- sound/soc/sh/rcar/ssiu.c | 178 +++++++++++++++++++-- 6 files changed, 255 insertions(+), 67 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index d92b705..8dfbe5b 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -363,6 +363,9 @@ Required properties: - 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. @@ -532,56 +535,55 @@ rcar_sound: sound@ec500000 { }; }; + 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>, <&audma0 0x15>, <&audma1 0x16>; - dma-names = "rx", "tx", "rxu", "txu"; + dmas = <&audma0 0x01>, <&audma1 0x02>; + dma-names = "rx", "tx"; }; ssi1: ssi-1 { interrupts = <0 371 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x03>, <&audma1 0x04>, <&audma0 0x49>, <&audma1 0x4a>; - dma-names = "rx", "tx", "rxu", "txu"; - }; - ssi2: ssi-2 { - interrupts = <0 372 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x05>, <&audma1 0x06>, <&audma0 0x63>, <&audma1 0x64>; - dma-names = "rx", "tx", "rxu", "txu"; - }; - ssi3: ssi-3 { - interrupts = <0 373 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x07>, <&audma1 0x08>, <&audma0 0x6f>, <&audma1 0x70>; - dma-names = "rx", "tx", "rxu", "txu"; - }; - ssi4: ssi-4 { - interrupts = <0 374 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x09>, <&audma1 0x0a>, <&audma0 0x71>, <&audma1 0x72>; - dma-names = "rx", "tx", "rxu", "txu"; - }; - ssi5: ssi-5 { - interrupts = <0 375 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x0b>, <&audma1 0x0c>, <&audma0 0x73>, <&audma1 0x74>; - dma-names = "rx", "tx", "rxu", "txu"; - }; - ssi6: ssi-6 { - interrupts = <0 376 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x0d>, <&audma1 0x0e>, <&audma0 0x75>, <&audma1 0x76>; - dma-names = "rx", "tx", "rxu", "txu"; - }; - ssi7: ssi-7 { - interrupts = <0 377 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x0f>, <&audma1 0x10>, <&audma0 0x79>, <&audma1 0x7a>; - dma-names = "rx", "tx", "rxu", "txu"; + dmas = <&audma0 0x03>, <&audma1 0x04>; + dma-names = "rx", "tx"; }; + + ... + ssi8: ssi-8 { interrupts = <0 378 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x11>, <&audma1 0x12>, <&audma0 0x7b>, <&audma1 0x7c>; - dma-names = "rx", "tx", "rxu", "txu"; + dmas = <&audma0 0x11>, <&audma1 0x12>; + dma-names = "rx", "tx"; }; ssi9: ssi-9 { interrupts = <0 379 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x13>, <&audma1 0x14>, <&audma0 0x7d>, <&audma1 0x7e>; - dma-names = "rx", "tx", "rxu", "txu"; + dmas = <&audma0 0x13>, <&audma1 0x14>; + dma-names = "rx", "tx"; }; }; diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 5e10012..fff0a3a 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1121,6 +1121,7 @@ static void __rsnd_dai_probe(struct rsnd_priv *priv, break; rsnd_parse_connect_ssi(rdai, playback, capture); + rsnd_parse_connect_ssiu(rdai, playback, capture); rsnd_parse_connect_src(rdai, playback, capture); rsnd_parse_connect_ctu(rdai, playback, capture); rsnd_parse_connect_mix(rdai, playback, capture); diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 5daa6c9..0324a5c 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -218,7 +218,7 @@ struct dma_chan *rsnd_dma_request_channel(struct device_node *of_node, int i = 0; for_each_child_of_node(of_node, np) { - if (i == rsnd_mod_id(mod) && (!chan)) + if (i == rsnd_mod_id_raw(mod) && (!chan)) chan = of_dma_request_slave_channel(np, name); i++; } @@ -344,14 +344,16 @@ static u32 rsnd_dmapp_get_id(struct rsnd_dai_stream *io, struct rsnd_mod *mod) { struct rsnd_mod *ssi = rsnd_io_to_mod_ssi(io); + struct rsnd_mod *ssiu = rsnd_io_to_mod_ssiu(io); struct rsnd_mod *src = rsnd_io_to_mod_src(io); struct rsnd_mod *dvc = rsnd_io_to_mod_dvc(io); const u8 *entry = NULL; int id = 255; int size = 0; - if (mod == ssi) { - int busif = rsnd_ssi_get_busif(io); + if ((mod == ssi) || + (mod == ssiu)) { + int busif = rsnd_mod_id_sub(ssiu); entry = gen2_id_table_ssiu; size = ARRAY_SIZE(gen2_id_table_ssiu); @@ -530,13 +532,14 @@ rsnd_gen2_dma_addr(struct rsnd_dai_stream *io, struct device *dev = rsnd_priv_to_dev(priv); phys_addr_t ssi_reg = rsnd_gen_get_phy_addr(priv, RSND_GEN2_SSI); phys_addr_t src_reg = rsnd_gen_get_phy_addr(priv, RSND_GEN2_SCU); - int is_ssi = !!(rsnd_io_to_mod_ssi(io) == mod); + int is_ssi = !!(rsnd_io_to_mod_ssi(io) == mod) || + !!(rsnd_io_to_mod_ssiu(io) == mod); int use_src = !!rsnd_io_to_mod_src(io); int use_cmd = !!rsnd_io_to_mod_dvc(io) || !!rsnd_io_to_mod_mix(io) || !!rsnd_io_to_mod_ctu(io); int id = rsnd_mod_id(mod); - int busif = rsnd_ssi_get_busif(io); + int busif = rsnd_mod_id_sub(rsnd_io_to_mod_ssiu(io)); struct dma_addr { dma_addr_t out_addr; dma_addr_t in_addr; @@ -620,7 +623,7 @@ static void rsnd_dma_of_path(struct rsnd_mod *this, struct rsnd_mod **mod_from, struct rsnd_mod **mod_to) { - struct rsnd_mod *ssi = rsnd_io_to_mod_ssi(io); + struct rsnd_mod *ssi; struct rsnd_mod *src = rsnd_io_to_mod_src(io); struct rsnd_mod *ctu = rsnd_io_to_mod_ctu(io); struct rsnd_mod *mix = rsnd_io_to_mod_mix(io); @@ -631,6 +634,28 @@ static void rsnd_dma_of_path(struct rsnd_mod *this, struct device *dev = rsnd_priv_to_dev(priv); int nr, i, idx; + /* + * It should use "rcar_sound,ssiu" on DT. + * But, we need to keep compatibility for old version. + * + * If it has "rcar_sound.ssiu", it will be used. + * If not, "rcar_sound.ssi" will be used. + * see + * rsnd_ssiu_dma_req() + * rsnd_ssi_dma_req() + */ + if (rsnd_ssiu_of_node(priv)) { + struct rsnd_mod *ssiu = rsnd_io_to_mod_ssiu(io); + + /* use SSIU */ + ssi = ssiu; + if (this == rsnd_io_to_mod_ssi(io)) + this = ssiu; + } else { + /* keep compatible, use SSI */ + ssi = rsnd_io_to_mod_ssi(io); + } + if (!ssi) return; diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 28bd90a..1875fd9 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -441,6 +441,7 @@ int rsnd_runtime_is_ssi_tdm(struct rsnd_dai_stream *io); of_get_child_by_name(rsnd_priv_to_dev(priv)->of_node, node) #define RSND_NODE_DAI "rcar_sound,dai" #define RSND_NODE_SSI "rcar_sound,ssi" +#define RSND_NODE_SSIU "rcar_sound,ssiu" #define RSND_NODE_SRC "rcar_sound,src" #define RSND_NODE_CTU "rcar_sound,ctu" #define RSND_NODE_MIX "rcar_sound,mix" @@ -715,7 +716,6 @@ void rsnd_ssi_remove(struct rsnd_priv *priv); struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id); int rsnd_ssi_is_dma_mode(struct rsnd_mod *mod); int rsnd_ssi_use_busif(struct rsnd_dai_stream *io); -int rsnd_ssi_get_busif(struct rsnd_dai_stream *io); u32 rsnd_ssi_multi_slaves_runtime(struct rsnd_dai_stream *io); #define RSND_SSI_HDMI_PORT0 0xf0 @@ -743,6 +743,10 @@ int rsnd_ssiu_attach(struct rsnd_dai_stream *io, struct rsnd_mod *mod); int rsnd_ssiu_probe(struct rsnd_priv *priv); void rsnd_ssiu_remove(struct rsnd_priv *priv); +void rsnd_parse_connect_ssiu(struct rsnd_dai *rdai, + struct device_node *playback, + struct device_node *capture); +#define rsnd_ssiu_of_node(priv) rsnd_parse_of_node(priv, RSND_NODE_SSIU) /* * R-Car SRC diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index a032819..95acc8b 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -150,11 +150,6 @@ int rsnd_ssi_use_busif(struct rsnd_dai_stream *io) return use_busif; } -int rsnd_ssi_get_busif(struct rsnd_dai_stream *io) -{ - return 0; /* BUSIF0 only for now */ -} - static void rsnd_ssi_status_clear(struct rsnd_mod *mod) { rsnd_mod_write(mod, SSISR, 0); @@ -761,7 +756,7 @@ static int rsnd_ssi_common_probe(struct rsnd_mod *mod, { struct device *dev = rsnd_priv_to_dev(priv); struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); - int ret; + int ret = 0; /* * SSIP/SSIU/IRQ are not needed on @@ -775,10 +770,6 @@ static int rsnd_ssi_common_probe(struct rsnd_mod *mod, * see rsnd_ssi_pcm_new() */ - ret = rsnd_ssiu_attach(io, mod); - if (ret < 0) - return ret; - /* * SSI might be called again as PIO fallback * It is easy to manual handling for IRQ request/free @@ -972,6 +963,17 @@ static struct dma_chan *rsnd_ssi_dma_req(struct rsnd_dai_stream *io, int is_play = rsnd_io_is_play(io); char *name; + /* + * It should use "rcar_sound,ssiu" on DT. + * But, we need to keep compatibility for old version. + * + * If it has "rcar_sound.ssiu", it will be used. + * If not, "rcar_sound.ssi" will be used. + * see + * rsnd_ssiu_dma_req() + * rsnd_dma_of_path() + */ + if (rsnd_ssi_use_busif(io)) name = is_play ? "rxu" : "txu"; else diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index ebcb826..20f7b04 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -12,6 +12,8 @@ struct rsnd_ssiu { struct rsnd_mod mod; u32 busif_status[8]; /* for BUSIF0 - BUSIF7 */ unsigned int usrcnt; + int id; + int id_sub; }; #define rsnd_ssiu_nr(priv) ((priv)->ssiu_nr) @@ -22,12 +24,29 @@ struct rsnd_ssiu { ((pos) = ((struct rsnd_ssiu *)(priv)->ssiu + i)); \ i++) +/* + * SSI Gen2 Gen3 + * 0 BUSIF0-3 BUSIF0-7 + * 1 BUSIF0-3 BUSIF0-7 + * 2 BUSIF0-3 BUSIF0-7 + * 3 BUSIF0 BUSIF0-7 + * 4 BUSIF0 BUSIF0-7 + * 5 BUSIF0 BUSIF0 + * 6 BUSIF0 BUSIF0 + * 7 BUSIF0 BUSIF0 + * 8 BUSIF0 BUSIF0 + * 9 BUSIF0-3 BUSIF0-7 + * total 22 52 + */ +static const int gen2_id[] = { 0, 4, 8, 12, 13, 14, 15, 16, 17, 18 }; +static const int gen3_id[] = { 0, 8, 16, 24, 32, 40, 41, 42, 43, 44 }; + static u32 *rsnd_ssiu_get_status(struct rsnd_mod *mod, struct rsnd_dai_stream *io, enum rsnd_mod_type type) { struct rsnd_ssiu *ssiu = rsnd_mod_to_ssiu(mod); - int busif = rsnd_ssi_get_busif(io); + int busif = rsnd_mod_id_sub(mod); return &ssiu->busif_status[busif]; } @@ -158,7 +177,7 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod, if (rsnd_ssi_use_busif(io)) { int id = rsnd_mod_id(mod); - int busif = rsnd_ssi_get_busif(io); + int busif = rsnd_mod_id_sub(mod); /* * FIXME @@ -255,7 +274,7 @@ static int rsnd_ssiu_start_gen2(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) { - int busif = rsnd_ssi_get_busif(io); + int busif = rsnd_mod_id_sub(mod); if (!rsnd_ssi_use_busif(io)) return 0; @@ -273,7 +292,7 @@ static int rsnd_ssiu_stop_gen2(struct rsnd_mod *mod, struct rsnd_priv *priv) { struct rsnd_ssiu *ssiu = rsnd_mod_to_ssiu(mod); - int busif = rsnd_ssi_get_busif(io); + int busif = rsnd_mod_id_sub(mod); if (!rsnd_ssi_use_busif(io)) return 0; @@ -289,8 +308,49 @@ static int rsnd_ssiu_stop_gen2(struct rsnd_mod *mod, return 0; } +static int rsnd_ssiu_id(struct rsnd_mod *mod) +{ + struct rsnd_ssiu *ssiu = rsnd_mod_to_ssiu(mod); + + /* see rsnd_ssiu_probe() */ + return ssiu->id; +} + +static int rsnd_ssiu_id_sub(struct rsnd_mod *mod) +{ + struct rsnd_ssiu *ssiu = rsnd_mod_to_ssiu(mod); + + /* see rsnd_ssiu_probe() */ + return ssiu->id_sub; +} + +static struct dma_chan *rsnd_ssiu_dma_req(struct rsnd_dai_stream *io, + struct rsnd_mod *mod) +{ + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + int is_play = rsnd_io_is_play(io); + char *name; + + /* + * It should use "rcar_sound,ssiu" on DT. + * But, we need to keep compatibility for old version. + * + * If it has "rcar_sound.ssiu", it will be used. + * If not, "rcar_sound.ssi" will be used. + * see + * rsnd_ssi_dma_req() + * rsnd_dma_of_path() + */ + + name = is_play ? "rx" : "tx"; + + return rsnd_dma_request_channel(rsnd_ssiu_of_node(priv), + mod, name); +} + static struct rsnd_mod_ops rsnd_ssiu_ops_gen2 = { .name = SSIU_NAME, + .dma_req = rsnd_ssiu_dma_req, .init = rsnd_ssiu_init_gen2, .start = rsnd_ssiu_start_gen2, .stop = rsnd_ssiu_stop_gen2, @@ -305,26 +365,83 @@ static struct rsnd_mod *rsnd_ssiu_mod_get(struct rsnd_priv *priv, int id) return rsnd_mod_get((struct rsnd_ssiu *)(priv->ssiu) + id); } -int rsnd_ssiu_attach(struct rsnd_dai_stream *io, - struct rsnd_mod *ssi_mod) +static void rsnd_parse_connect_ssiu_compatible(struct rsnd_priv *priv, + struct rsnd_dai_stream *io) +{ + struct rsnd_mod *ssi_mod = rsnd_io_to_mod_ssi(io); + struct rsnd_mod *mod; + struct rsnd_ssiu *ssiu; + int i; + + if (!ssi_mod) + return; + + /* select BUSIF0 */ + for_each_rsnd_ssiu(ssiu, priv, i) { + mod = rsnd_mod_get(ssiu); + + if ((rsnd_mod_id(ssi_mod) == rsnd_ssiu_id(mod)) && + (rsnd_mod_id_sub(mod) == 0)) + rsnd_dai_connect(mod, io, mod->type); + } +} + +void rsnd_parse_connect_ssiu(struct rsnd_dai *rdai, + struct device_node *playback, + struct device_node *capture) { - struct rsnd_priv *priv = rsnd_io_to_priv(io); - struct rsnd_mod *mod = rsnd_ssiu_mod_get(priv, rsnd_mod_id(ssi_mod)); + struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); + struct device_node *node = rsnd_ssiu_of_node(priv); + struct device_node *np; + struct rsnd_mod *mod; + struct rsnd_dai_stream *io_p = &rdai->playback; + struct rsnd_dai_stream *io_c = &rdai->capture; + int i; - rsnd_mod_confirm_ssi(ssi_mod); + /* use rcar_sound,ssiu if exist */ + if (node) { + i = 0; + for_each_child_of_node(node, np) { + mod = rsnd_ssiu_mod_get(priv, i); + if (np == playback) + rsnd_dai_connect(mod, io_p, mod->type); + if (np == capture) + rsnd_dai_connect(mod, io_c, mod->type); + i++; + } - return rsnd_dai_connect(mod, io, mod->type); + of_node_put(node); + } + + /* Keep DT compatibility */ + if (!rsnd_io_to_mod_ssiu(io_p)) + rsnd_parse_connect_ssiu_compatible(priv, io_p); + if (!rsnd_io_to_mod_ssiu(io_c)) + rsnd_parse_connect_ssiu_compatible(priv, io_c); } int rsnd_ssiu_probe(struct rsnd_priv *priv) { struct device *dev = rsnd_priv_to_dev(priv); + struct device_node *node; struct rsnd_ssiu *ssiu; struct rsnd_mod_ops *ops; + const int *list = NULL; int i, nr, ret; - /* same number to SSI */ - nr = priv->ssi_nr; + /* + * Keep DT compatibility. + * if it has "rcar_sound,ssiu", use it. + * if not, use "rcar_sound,ssi" + * see + * rsnd_ssiu_bufsif_to_id() + */ + node = rsnd_ssiu_of_node(priv); + if (node) + nr = of_get_child_count(node); + else + nr = priv->ssi_nr; + ssiu = devm_kcalloc(dev, nr, sizeof(*ssiu), GFP_KERNEL); if (!ssiu) return -ENOMEM; @@ -337,7 +454,44 @@ int rsnd_ssiu_probe(struct rsnd_priv *priv) else ops = &rsnd_ssiu_ops_gen2; + /* Keep compatibility */ + nr = 0; + if ((node) && + (ops == &rsnd_ssiu_ops_gen2)) { + ops->id = rsnd_ssiu_id; + ops->id_sub = rsnd_ssiu_id_sub; + + if (rsnd_is_gen2(priv)) { + list = gen2_id; + nr = ARRAY_SIZE(gen2_id); + } else if (rsnd_is_gen3(priv)) { + list = gen3_id; + nr = ARRAY_SIZE(gen3_id); + } else { + dev_err(dev, "unknown SSIU\n"); + return -ENODEV; + } + } + for_each_rsnd_ssiu(ssiu, priv, i) { + if (node) { + int j; + + /* + * see + * rsnd_ssiu_get_id() + * rsnd_ssiu_get_id_sub() + */ + for (j = 0; j < nr; j++) { + if (list[j] > i) + break; + ssiu->id = j; + ssiu->id_sub = i - list[ssiu->id]; + } + } else { + ssiu->id = i; + } + ret = rsnd_mod_init(priv, rsnd_mod_get(ssiu), ops, NULL, RSND_MOD_SSIU, i); if (ret) -- 2.7.4

2 2

Re: [alsa-devel] [EXTERNAL] Re: [PATCH 1/2] ASoC: cs35l36: Add support for Cirrus CS35L36 Amplifier
by Schulman, James 15 Nov '18

15 Nov '18

On Tue, 13 Nov 2018, Mark Brown wrote: > On Tue, Nov 13, 2018 at 12:49:16PM -0600, James Schulman wrote: > >> +++ b/sound/soc/codecs/cs35l36-tables.c >> @@ -0,0 +1,315 @@ >> +/* >> + * cs35l36-tables.c -- CS35L36 ALSA SoC audio driver >> + * >> + * Copyright 2018 Cirrus Logic, Inc. > > Please use SPDX headers. > Will do. >> + SOC_SINGLE_TLV("AMP PCM Gain", CS35L36_AMP_GAIN_CTRL, 5, 0x13, 0, >> + amp_gain_tlv), > > All volume controls should end in Volume, see control-names.rst. > >> + regmap_read(cs35l36->regmap, CS35L36_INT4_RAW_STATUS, &reg); >> + if (reg & CS35L36_PLL_UNLOCK_MASK) >> + dev_crit(cs35l36->dev, "PLL Unlocked\n"); > > WARN_ON_ONCE? > Will do. >> + } >> + return 0; >> +} >> +static const char * const cs35l36_chan_text[] = { > > Missing blank line. > Will do. >> +static const char * const cs35l36_boost_text[] = { >> + "On", >> + "Off", >> +}; >> + >> +static SOC_ENUM_SINGLE_DECL(boost_enum, SND_SOC_NOPM, 0, >> + cs35l36_boost_text); >> + >> +static const struct snd_kcontrol_new cs35l36_boost_mux[] = { >> + SOC_DAPM_ENUM("Boost Enable", boost_enum), >> +}; > > Simple on/off conttrols should be _SINGLE controls with Switch at the > end of their name. It's not super clear why this is in DAPM as a widget > at all, the routing around it is: > > + {"BOOST Mux", "On", "Channel Mux"}, > + {"CLASS H", NULL, "BOOST Mux"}, > > which is a bit perplexing. Possibly this should be handled in the event > for the main amplifier? > We used _ENUM instead of _SINGLE because we wanted the default value to be "On" instead of "Off". We don't want the default assumption to have boost turned off. However, we're happy to change this if you would rather have consitency with the _SINGLE controls. >> + if (cs35l36->sclk > 6000000) { >> + fs1_val = 3 * 4 + 4; >> + fs2_val = 8 * 4 + 4; >> + } > > These are just constants, why write out the operations? > Will do. >> + >> + if (cs35l36->sclk <= 6000000) { > > if/else? > Will do. >> + /* >> + * Rev B0 has 2 versions >> + * L36 is 10V >> + * L37 is 12V > > \o/ > Yes we had 2 silicon spins for this revision >_< is there anything about this comment you want us to change? >> + } >> + >> + return IRQ_HANDLED; > > This unconditionally reports that it handled the interrupt, even if it > didn't. This will stop the interrupt core handling for faulty interrupt > lines working and will disrupt any support that gets added for handling > shared threaded interrupts. > Earlier in the handler, we have a check to see if any unmasked bits are set. If not we return IRQ_NONE. I figured this would be enough to ensure we don't unconditionally report IRQ_HANDLED. >> + if (pdata) >> + cs35l36->pdata = *pdata; >> + else { > > Use { } on both sides of the if (see coding-style.rst)? > Will do. >> + cs35l36->reset_gpio = devm_gpiod_get_optional(dev, "reset", >> + GPIOD_OUT_LOW); >> + if (IS_ERR(cs35l36->reset_gpio)) { >> + ret = PTR_ERR(cs35l36->reset_gpio); >> + cs35l36->reset_gpio = NULL; >> + if (ret == -EBUSY) >> + dev_info(dev, >> + "Reset line busy, assuming shared reset\n"); >> + else { >> + dev_err(dev, "Failed to get reset GPIO: %d\n", ret); >> + goto err; >> + } >> + } > > This doesn't handle -EPROBE_DEFER. > I think we will add an err_disable_regs jump statement to ensure we don't try to use the uninitialized reset_gpio handle. Once we do that, I think this will handle -EPROBE_DEFER cleanly? Thanks for the comments Mark. - James

2 3

[alsa-devel] [PATCH] ASoC: dapm: Recalculate audio map forcely when card instantiated
by Tzung-Bi Shih 15 Nov '18

15 Nov '18

Audio map are possible in wrong state before card->instantiated has been set to true. Imaging the following examples: time 1: at the beginning in:-1 in:-1 in:-1 in:-1 out:-1 out:-1 out:-1 out:-1 SIGGEN A B Spk time 2: after someone called snd_soc_dapm_new_widgets() (e.g. create_fill_widget_route_map() in sound/soc/codecs/hdac_hdmi.c) in:1 in:0 in:0 in:0 out:0 out:0 out:0 out:1 SIGGEN A B Spk time 3: routes added in:1 in:0 in:0 in:0 out:0 out:0 out:0 out:1 SIGGEN -----> A -----> B ---> Spk In the end, the path should be powered on but it did not. At time 3, "in" of SIGGEN and "out" of Spk did not propagate to their neighbors because snd_soc_dapm_add_path() will not invalidate the paths if the card has not instantiated (i.e. card->instantiated is false). To correct the state of audio map, recalculate the whole map forcely. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- sound/soc/soc-core.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6ddcf12bc030..b29d0f65611e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2131,6 +2131,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) } card->instantiated = 1; + dapm_mark_endpoints_dirty(card); snd_soc_dapm_sync(&card->dapm); mutex_unlock(&card->mutex); mutex_unlock(&client_mutex); -- 2.19.1.930.g4563a0d9d0-goog

2 1

[alsa-devel] Applied "ASoC: pcm3060: Add DT property for single-ended output" to the asoc tree
by Mark Brown 15 Nov '18

15 Nov '18

The patch ASoC: pcm3060: Add DT property for single-ended output has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 9d4b17efa51f3bb0918021090c09b0013039df00 Mon Sep 17 00:00:00 2001 From: Kirill Marinushkin <kmarinushkin(a)birdec.tech> Date: Mon, 12 Nov 2018 08:08:33 +0100 Subject: [PATCH] ASoC: pcm3060: Add DT property for single-ended output DAC output may be differential (default) or single-ended. Signed-off-by: Kirill Marinushkin <kmarinushkin(a)birdec.tech> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/pcm3060.c | 14 ++++++++++++++ sound/soc/codecs/pcm3060.h | 1 + 2 files changed, 15 insertions(+) diff --git a/sound/soc/codecs/pcm3060.c b/sound/soc/codecs/pcm3060.c index 494d9d662be8..5f479aa61097 100644 --- a/sound/soc/codecs/pcm3060.c +++ b/sound/soc/codecs/pcm3060.c @@ -274,9 +274,23 @@ EXPORT_SYMBOL(pcm3060_regmap); /* device */ +static void pcm3060_parse_dt(const struct device_node *np, + struct pcm3060_priv *priv) +{ + priv->out_se = of_property_read_bool(np, "ti,out-single-ended"); +} + int pcm3060_probe(struct device *dev) { int rc; + struct pcm3060_priv *priv = dev_get_drvdata(dev); + + if (dev->of_node) + pcm3060_parse_dt(dev->of_node, priv); + + if (priv->out_se) + regmap_update_bits(priv->regmap, PCM3060_REG64, + PCM3060_REG_SE, PCM3060_REG_SE); rc = devm_snd_soc_register_component(dev, &pcm3060_soc_comp_driver, pcm3060_dai, diff --git a/sound/soc/codecs/pcm3060.h b/sound/soc/codecs/pcm3060.h index fd89a68aa8a7..c895cf40ee10 100644 --- a/sound/soc/codecs/pcm3060.h +++ b/sound/soc/codecs/pcm3060.h @@ -25,6 +25,7 @@ struct pcm3060_priv_dai { struct pcm3060_priv { struct regmap *regmap; struct pcm3060_priv_dai dai[PCM3060_DAI_IDS_NUM]; + u8 out_se: 1; }; int pcm3060_probe(struct device *dev); -- 2.19.1

1 0

[alsa-devel] Applied "ASoC: amd: fix memory leak of i2s_data on error return" to the asoc tree
by Mark Brown 15 Nov '18

15 Nov '18

The patch ASoC: amd: fix memory leak of i2s_data on error return has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 46dce4042659755a2b9753369d0c2d8ca33e35cf Mon Sep 17 00:00:00 2001 From: Colin Ian King <colin.king(a)canonical.com> Date: Wed, 14 Nov 2018 21:31:48 +0000 Subject: [PATCH] ASoC: amd: fix memory leak of i2s_data on error return Currently when snd_pcm_hw_constraint_integer fails there is a memory leak of i2s_data on the error return path. Fix this by kfree'ing i2s_data before returning. Detected by CoverityScan, CID#1475479 ("Resource leak") Fixes: 0b87d6bcd648 ("ASoC: amd: add acp3x pcm driver dma ops") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 2e61cef00f41..b16cdd187139 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -309,6 +309,7 @@ static int acp3x_dma_open(struct snd_pcm_substream *substream) SNDRV_PCM_HW_PARAM_PERIODS); if (ret < 0) { dev_err(component->dev, "set integer constraint failed\n"); + kfree(i2s_data); return ret; } -- 2.19.1

1 0

[alsa-devel] [PATCH] ASoC: omap-abe-twl6040: Fix missing audio card caused by deferred probing
by Peter Ujfalusi 15 Nov '18

15 Nov '18

The machine driver fails to probe in next-20181113 with: [ 2.539093] omap-abe-twl6040 sound: ASoC: CODEC DAI twl6040-legacy not registered [ 2.546630] omap-abe-twl6040 sound: devm_snd_soc_register_card() failed: -517 ... [ 3.693206] omap-abe-twl6040 sound: ASoC: Both platform name/of_node are set for TWL6040 [ 3.701446] omap-abe-twl6040 sound: ASoC: failed to init link TWL6040 [ 3.708007] omap-abe-twl6040 sound: devm_snd_soc_register_card() failed: -22 [ 3.715148] omap-abe-twl6040: probe of sound failed with error -22 Bisect pointed to a merge commit: first bad commit: [0f688ab20a540aafa984c5dbd68a71debebf4d7f] Merge remote-tracking branch 'net-next/master' and a diff between a working kernel does not reveal anything which would explain the change in behavior. Further investigation showed that on the second try of loading fails because the dai_link->platform is no longer NULL and it might be pointing to uninitialized memory. The fix is to move the snd_soc_dai_link and snd_soc_card inside of the abe_twl6040 struct, which is dynamically allocated every time the driver probes. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/omap-abe-twl6040.c | 67 +++++++++++++------------------ 1 file changed, 29 insertions(+), 38 deletions(-) diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index d5ae9eb8c756..fed45b41f9d3 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -36,6 +36,8 @@ #include "../codecs/twl6040.h" struct abe_twl6040 { + struct snd_soc_card card; + struct snd_soc_dai_link dai_links[2]; int jack_detection; /* board can detect jack events */ int mclk_freq; /* MCLK frequency speed for twl6040 */ }; @@ -208,40 +210,10 @@ static int omap_abe_dmic_init(struct snd_soc_pcm_runtime *rtd) ARRAY_SIZE(dmic_audio_map)); } -/* Digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link abe_twl6040_dai_links[] = { - { - .name = "TWL6040", - .stream_name = "TWL6040", - .codec_dai_name = "twl6040-legacy", - .codec_name = "twl6040-codec", - .init = omap_abe_twl6040_init, - .ops = &omap_abe_ops, - }, - { - .name = "DMIC", - .stream_name = "DMIC Capture", - .codec_dai_name = "dmic-hifi", - .codec_name = "dmic-codec", - .init = omap_abe_dmic_init, - .ops = &omap_abe_dmic_ops, - }, -}; - -/* Audio machine driver */ -static struct snd_soc_card omap_abe_card = { - .owner = THIS_MODULE, - - .dapm_widgets = twl6040_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets), - .dapm_routes = audio_map, - .num_dapm_routes = ARRAY_SIZE(audio_map), -}; - static int omap_abe_probe(struct platform_device *pdev) { struct device_node *node = pdev->dev.of_node; - struct snd_soc_card *card = &omap_abe_card; + struct snd_soc_card *card; struct device_node *dai_node; struct abe_twl6040 *priv; int num_links = 0; @@ -252,12 +224,18 @@ static int omap_abe_probe(struct platform_device *pdev) return -ENODEV; } - card->dev = &pdev->dev; - priv = devm_kzalloc(&pdev->dev, sizeof(struct abe_twl6040), GFP_KERNEL); if (priv == NULL) return -ENOMEM; + card = &priv->card; + card->dev = &pdev->dev; + card->owner = THIS_MODULE; + card->dapm_widgets = twl6040_dapm_widgets; + card->num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets); + card->dapm_routes = audio_map; + card->num_dapm_routes = ARRAY_SIZE(audio_map); + if (snd_soc_of_parse_card_name(card, "ti,model")) { dev_err(&pdev->dev, "Card name is not provided\n"); return -ENODEV; @@ -274,14 +252,27 @@ static int omap_abe_probe(struct platform_device *pdev) dev_err(&pdev->dev, "McPDM node is not provided\n"); return -EINVAL; } - abe_twl6040_dai_links[0].cpu_of_node = dai_node; - abe_twl6040_dai_links[0].platform_of_node = dai_node; + + priv->dai_links[0].name = "DMIC"; + priv->dai_links[0].stream_name = "TWL6040"; + priv->dai_links[0].cpu_of_node = dai_node; + priv->dai_links[0].platform_of_node = dai_node; + priv->dai_links[0].codec_dai_name = "twl6040-legacy"; + priv->dai_links[0].codec_name = "twl6040-codec"; + priv->dai_links[0].init = omap_abe_twl6040_init; + priv->dai_links[0].ops = &omap_abe_ops; dai_node = of_parse_phandle(node, "ti,dmic", 0); if (dai_node) { num_links = 2; - abe_twl6040_dai_links[1].cpu_of_node = dai_node; - abe_twl6040_dai_links[1].platform_of_node = dai_node; + priv->dai_links[1].name = "TWL6040"; + priv->dai_links[1].stream_name = "DMIC Capture"; + priv->dai_links[1].cpu_of_node = dai_node; + priv->dai_links[1].platform_of_node = dai_node; + priv->dai_links[1].codec_dai_name = "dmic-hifi"; + priv->dai_links[1].codec_name = "dmic-codec"; + priv->dai_links[1].init = omap_abe_dmic_init; + priv->dai_links[1].ops = &omap_abe_dmic_ops; } else { num_links = 1; } @@ -300,7 +291,7 @@ static int omap_abe_probe(struct platform_device *pdev) return -ENODEV; } - card->dai_link = abe_twl6040_dai_links; + card->dai_link = priv->dai_links; card->num_links = num_links; snd_soc_card_set_drvdata(card, priv); -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 1

[alsa-devel] [PATCH][next] ASoC: amd: fix memory leak of i2s_data on error return
by Colin King 14 Nov '18

14 Nov '18

From: Colin Ian King <colin.king(a)canonical.com> Currently when snd_pcm_hw_constraint_integer fails there is a memory leak of i2s_data on the error return path. Fix this by kfree'ing i2s_data before returning. Detected by CoverityScan, CID#1475479 ("Resource leak") Fixes: 0b87d6bcd648 ("ASoC: amd: add acp3x pcm driver dma ops") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 2e61cef00f41..b16cdd187139 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -309,6 +309,7 @@ static int acp3x_dma_open(struct snd_pcm_substream *substream) SNDRV_PCM_HW_PARAM_PERIODS); if (ret < 0) { dev_err(component->dev, "set integer constraint failed\n"); + kfree(i2s_data); return ret; } -- 2.19.1

1 0

Re: [alsa-devel] linux-next: Tree for Nov 14 (sound/soc/amd/raven/)
by Randy Dunlap 14 Nov '18

14 Nov '18

On 11/13/18 9:26 PM, Stephen Rothwell wrote: > Hi all, > > Changes since 20181113: > on i386 or x86_64: when CONFIG_PCI is not enabled. ../sound/soc/amd/raven/pci-acp3x.c: In function 'snd_acp3x_probe': ../sound/soc/amd/raven/pci-acp3x.c:59:2: error: implicit declaration of function 'pci_enable_msi' [-Werror=implicit-function-declaration] ret = pci_enable_msi(pci); ^ ../sound/soc/amd/raven/pci-acp3x.c:124:2: error: implicit declaration of function 'pci_disable_msi' [-Werror=implicit-function-declaration] pci_disable_msi(pci); ^ ../sound/soc/amd/raven/pci-acp3x.c: At top level: ../sound/soc/amd/raven/pci-acp3x.c:161:1: warning: data definition has no type or storage class [enabled by default] module_pci_driver(acp3x_driver); ^ ../sound/soc/amd/raven/pci-acp3x.c:161:1: error: type defaults to 'int' in declaration of 'module_pci_driver' [-Werror=implicit-int] ../sound/soc/amd/raven/pci-acp3x.c:161:1: warning: parameter names (without types) in function declaration [enabled by default] ../sound/soc/amd/raven/pci-acp3x.c:154:26: warning: 'acp3x_driver' defined but not used [-Wunused-variable] static struct pci_driver acp3x_driver = { ^ config SND_SOC_AMD_ACP3x tristate "AMD Audio Coprocessor-v3.x support" help This option enables ACP v3.x I2S support on AMD platform Needs "depends on PCI"?? -- ~Randy

1 0

[alsa-devel] alsa-utils: axfer: rewrite aplay, adding 'timer-based scheduling' option
by GitHub pull_request - opened 14 Nov '18

14 Nov '18

alsa-project/alsa-utils pull request #3 was opened from takaswie: Hi, This patchset is an updated version of my previous RFCv3, and for merging to upstream. [alsa-devel] [RFCv3][PATCH 00/39] axfer: rewrite aplay, adding 'timer-based scheduling' option http://mailman.alsa-project.org/pipermail/alsa-devel/2017-October/125961.ht… As I noted in RFCv3, some patches are dropped in this patchset, due to some technical issues. In detail, please refer to RFCv3. Changes from RFCv3: - rename scheduling option to 'sched-model' - improve waiter implementation and add support for select(2) - categorize supported options as common and backend-dependent - release alsa-lib configuration space at the end of runtime - valgrind test and fix some memory leaks - add 'frame-cache' internal interface to share implementation by several backends. - add 'libffado' backend just for technical preview (linkage of this library brings problems. Please refet to patch comment.) I apologize but in this time I don't prepare any help document and manuals as well as RFCv3. Let me put them to a part of task for this development period. Regards Takashi Sakamoto (35): axfer: add an entry point for this command axfer: add a sub-command to print list of PCMs/devices axfer: add a common interface to handle a file with audio-specific data format axfer: add support for a container of Microsoft/IBM RIFF/Wave format axfer: add support for a container of Sparc AU format axfer: add support for a container of Creative Tech. voice format axfer: add support for a container of raw data axfer: add unit test for container interface axfer: add a common interface to align data frames on different layout axfer: add support for a mapper for single target axfer: add support for a mapper for multiple target axfer: add a unit test for mapper interface axfer: add a common interface to transfer data frames axfer: add a parser for command-line options axfer: add support to transfer data frames by alsa-lib PCM APIs axfer: add support for blocking data transmission operation of alsa-lib PCM API axfer: add a sub-command to transfer data frames axfer: add informative output and an option to suppress it axfer: add an option to dump available hardware parameters axfer: add options related to duration and obsolete '--max-file-size' option axfer: add an option to finish transmission at XRUN axfer: add support for non-blocking operation axfer: add support for MMAP PCM operation axfer: add an option to suppress event waiting axfer: add options for buffer arrangement axfer: add options for software parameters of PCM substream axfer: add options for plugins in alsa-lib axfer: add a common interface of waiter for I/O event notification axfer: add an option for waiter type axfer: add an implementation of waiter for poll(2) axfer: add an implementation of waiter for select(2) axfer: add an implementation of waiter for epoll(7) axfer: add support for timer-based scheduling model with MMAP operation axfer: obsolete some unimplemented options axfer: add support for libffado transmission backend ``` Makefile.am | 2 +- axfer/Makefile.am | 62 +++ axfer/container-au.c | 203 +++++++ axfer/container-raw.c | 66 +++ axfer/container-riff-wave.c | 575 +++++++++++++++++++ axfer/container-voc.c | 833 +++++++++++++++++++++++++++ axfer/container.c | 457 +++++++++++++++ axfer/container.h | 126 +++++ axfer/frame-cache.c | 111 ++++ axfer/frame-cache.h | 47 ++ axfer/main.c | 229 ++++++++ axfer/mapper-multiple.c | 259 +++++++++ axfer/mapper-single.c | 191 +++++++ axfer/mapper.c | 150 +++++ axfer/mapper.h | 87 +++ axfer/misc.h | 19 + axfer/subcmd-list.c | 234 ++++++++ axfer/subcmd-transfer.c | 467 ++++++++++++++++ axfer/subcmd.h | 18 + axfer/test/Makefile.am | 31 ++ axfer/test/container-test.c | 299 ++++++++++ axfer/test/generator.c | 260 +++++++++ axfer/test/generator.h | 47 ++ axfer/test/mapper-test.c | 491 ++++++++++++++++ axfer/waiter-epoll.c | 107 ++++ axfer/waiter-poll.c | 45 ++ axfer/waiter-select.c | 100 ++++ axfer/waiter.c | 104 ++++ axfer/waiter.h | 61 ++ axfer/xfer-libasound-irq-mmap.c | 282 ++++++++++ axfer/xfer-libasound-irq-rw.c | 427 ++++++++++++++ axfer/xfer-libasound-timer-mmap.c | 455 +++++++++++++++ axfer/xfer-libasound.c | 898 ++++++++++++++++++++++++++++++ axfer/xfer-libasound.h | 92 +++ axfer/xfer-libffado.c | 555 ++++++++++++++++++ axfer/xfer-options.c | 589 ++++++++++++++++++++ axfer/xfer.c | 254 +++++++++ axfer/xfer.h | 115 ++++ configure.ac | 15 +- 39 files changed, 9361 insertions(+), 2 deletions(-) create mode 100644 axfer/Makefile.am create mode 100644 axfer/container-au.c create mode 100644 axfer/container-raw.c create mode 100644 axfer/container-riff-wave.c create mode 100644 axfer/container-voc.c create mode 100644 axfer/container.c create mode 100644 axfer/container.h create mode 100644 axfer/frame-cache.c create mode 100644 axfer/frame-cache.h create mode 100644 axfer/main.c create mode 100644 axfer/mapper-multiple.c create mode 100644 axfer/mapper-single.c create mode 100644 axfer/mapper.c create mode 100644 axfer/mapper.h create mode 100644 axfer/misc.h create mode 100644 axfer/subcmd-list.c create mode 100644 axfer/subcmd-transfer.c create mode 100644 axfer/subcmd.h create mode 100644 axfer/test/Makefile.am create mode 100644 axfer/test/container-test.c create mode 100644 axfer/test/generator.c create mode 100644 axfer/test/generator.h create mode 100644 axfer/test/mapper-test.c create mode 100644 axfer/waiter-epoll.c create mode 100644 axfer/waiter-poll.c create mode 100644 axfer/waiter-select.c create mode 100644 axfer/waiter.c create mode 100644 axfer/waiter.h create mode 100644 axfer/xfer-libasound-irq-mmap.c create mode 100644 axfer/xfer-libasound-irq-rw.c create mode 100644 axfer/xfer-libasound-timer-mmap.c create mode 100644 axfer/xfer-libasound.c create mode 100644 axfer/xfer-libasound.h create mode 100644 axfer/xfer-libffado.c create mode 100644 axfer/xfer-options.c create mode 100644 axfer/xfer.c create mode 100644 axfer/xfer.h ``` Request URL : https://github.com/alsa-project/alsa-utils/pull/3 Patch URL : https://github.com/alsa-project/alsa-utils/pull/3.patch Repository URL: https://github.com/alsa-project/alsa-utils

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[alsa-devel] GitHub - alsa-project - repositories
by Jaroslav Kysela 14 Nov '18

14 Nov '18

Hi all, URL: https://github.com/alsa-project I finally finished the first phase of the integration with GitHub. All repositories are now on GitHub and all repositories are mirrored to alsa-project.org. The github repositories are master (developers should push changes only to those repositories). If you refer the repository as the code source (for packaging or so), please, keep to use the repositories on alsa-project.org (in case when we migrate to other service in future like GitLab or so). The mirroring is realtime, so the changes should be visible on git.alsa-project.org in few seconds after the push. I invited few people to the GitHub team and actually Takashi has full access to all repos, Vinod Koul should have the write access to tinycompress and Takashi Sakamoto should have the write access to alsa-gi. If I omitted someone (or someone is not on github), please, let me know (and register before). Because github adds possibility for the pull requests and issue tracking, I added notifications for them to this (alsa-devel) mailing list. The short notification should be sent when an pull request or an issue is opened or changed. I think that it would be best to handle this per request, so the developer can ask to resend the patch to the mailing list for the wider review or just push the change with the signing. I activated Travis CI for alsa-lib and alsa-utils and I will add other repos soon, too. URL: https://travis-ci.org/alsa-project If you have some ideas which other github applications can be used to improve the code maintenance, let me know. I will probably play with the coverity checker (http://scan.coverity.com), too. Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

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