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

[alsa-devel] [PATCH RFC] ASoC: simple-card: Enable and disable DAI clocks as needed
by Jyri Sarha 15 Jan '15

15 Jan '15

Call clk_prepare_enable() and clk_disable_unprepare() for cpu dai clock and codec dai clock in dai statup and shutdown callbacks. This to make sure the related clock are enabled when the audio device is used. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- This patch is needed for Beaglebone-back HDMI audio. There is no direct dependency between this patch and the other (dts and DRM) changes needed for the HDMI audio so these changes can be merged independently. There are also some other TI EVMs that require this functionality if they are to be used with simple-card. .../devicetree/bindings/sound/simple-card.txt | 5 +++ include/sound/simple_card.h | 1 + sound/soc/generic/simple-card.c | 34 ++++++++++++++++++++ 3 files changed, 40 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt index c3cba60..73bf314 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ b/Documentation/devicetree/bindings/sound/simple-card.txt @@ -75,6 +75,11 @@ Optional CPU/CODEC subnodes properties: 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(). Example 1 - single DAI link: diff --git a/include/sound/simple_card.h b/include/sound/simple_card.h index 9b0ac77..1255ddb 100644 --- a/include/sound/simple_card.h +++ b/include/sound/simple_card.h @@ -20,6 +20,7 @@ struct asoc_simple_dai { unsigned int sysclk; int slots; int slot_width; + struct clk *clk; }; struct asoc_simple_card_info { diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index fb9240f..cb3998d 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -39,6 +39,37 @@ struct simple_card_data { #define simple_priv_to_link(priv, i) ((priv)->snd_card.dai_link + i) #define simple_priv_to_props(priv, i) ((priv)->dai_props + i) +static int asoc_simple_card_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); + struct simple_dai_props *dai_props = + &priv->dai_props[rtd - rtd->card->rtd]; + int ret; + + ret = clk_prepare_enable(dai_props->cpu_dai.clk); + if (ret) + return ret; + + ret = clk_prepare_enable(dai_props->codec_dai.clk); + if (ret) + clk_disable_unprepare(dai_props->cpu_dai.clk); + + return ret; +} + +static void asoc_simple_card_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); + struct simple_dai_props *dai_props = + &priv->dai_props[rtd - rtd->card->rtd]; + + clk_disable_unprepare(dai_props->cpu_dai.clk); + + clk_disable_unprepare(dai_props->codec_dai.clk); +} + static int asoc_simple_card_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -58,6 +89,8 @@ static int asoc_simple_card_hw_params(struct snd_pcm_substream *substream, } static struct snd_soc_ops asoc_simple_card_ops = { + .startup = asoc_simple_card_startup, + .shutdown = asoc_simple_card_shutdown, .hw_params = asoc_simple_card_hw_params, }; @@ -219,6 +252,7 @@ asoc_simple_card_sub_parse_of(struct device_node *np, } dai->sysclk = clk_get_rate(clk); + dai->clk = clk; } else if (!of_property_read_u32(np, "system-clock-frequency", &val)) { dai->sysclk = val; } else { -- 1.7.9.5

2 1

[alsa-devel] [PATCH 0/16] ASoC: rsnd: cleanup patches
by Kuninori Morimoto 15 Jan '15

15 Jan '15

Hi Mark These are cleanup patches for Renesas R-Car sound driver. I noticed current rsnd driver has un-used function, no-meaning parameter, dirty code, etc... These are cleanup patches for it. Kuninori Morimoto (16): ASoC: rsnd: remove unsed .rate ASoC: rsnd: tidyup rsnd_io_to_xxx() macro declaration position ASoC: rsnd: rsnd_mod controls clock ASoC: rsnd: rename rsnd_dai_get() to rsnd_rdai_get() ASoC: rsnd: rename rsnd_dai_is_clk_master() to rsnd_rdai_is_clk_master() ASoC: rsnd: remove unused rsnd_dai_id() ASoC: rsnd: remove rsnd_dai_get_platform_info() macro ASoC: rsnd: add rsnd_io_to_rdai() ASoC: rsnd: replace rsnd_dai_is_play() to rsnd_io_is_play() ASoC: rsnd: remove unsed rdai parameter from each function ASoC: rsnd: replace rdai to priv in each function ASoC: rsnd: save priv in struct rsnd_dai ASoC: rsnd: error meesage indicates its port ASoC: rsnd: ignore DIRQ when DMA transfer ASoC: rsnd: tidyup DMA start/stop timing on SSI ASoC: rsnd: tidyup DVC/SRC/SSI order sound/soc/sh/rcar/adg.c | 14 +++---- sound/soc/sh/rcar/core.c | 91 +++++++++++++++++------------------------ sound/soc/sh/rcar/dvc.c | 48 ++++++++++------------ sound/soc/sh/rcar/rsnd.h | 68 ++++++++++++++----------------- sound/soc/sh/rcar/src.c | 94 +++++++++++++++++++------------------------ sound/soc/sh/rcar/ssi.c | 100 +++++++++++++++++++++------------------------- 6 files changed, 178 insertions(+), 237 deletions(-) Best regards --- Kuninori Morimoto

2 17

[alsa-devel] [PATCH] ASoC: rt286: set the same format for dac and adc
by Bard Liao 15 Jan '15

15 Jan '15

There is only one I2S I/F, AD/DA path must operate to the same format. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt286.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 2cd4fe4..1d1c7f8 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -861,10 +861,8 @@ static int rt286_hw_params(struct snd_pcm_substream *substream, RT286_I2S_CTRL1, 0x0018, d_len_code << 3); dev_dbg(codec->dev, "format val = 0x%x\n", val); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - snd_soc_update_bits(codec, RT286_DAC_FORMAT, 0x407f, val); - else - snd_soc_update_bits(codec, RT286_ADC_FORMAT, 0x407f, val); + snd_soc_update_bits(codec, RT286_DAC_FORMAT, 0x407f, val); + snd_soc_update_bits(codec, RT286_ADC_FORMAT, 0x407f, val); return 0; } -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] unload Audio drivers while playback stream is active case kernel crash
by Wang, Jiada (ESD) 15 Jan '15

15 Jan '15

Hi Community I am using i.MX6Q sabreSD board, which have imx_wm892 machine driver, wm8962 codec and SSI CPU DAI, I got Kernel crash when unloading audio drivers (playback stream is active) modprobe -r snd_soc_imx_wm8962 modprobe -r snd_soc_fsl_ssi modprobe -r snd_soc_wm8962 [ 208.666868] Unable to handle kernel paging request at virtc [ 208.674110] pgd = 80004000 [ 208.676867] [7f06541c] *pgd=4c334811, *pte=00000000, *ppte=00000000 [ 208.683211] Internal error: Oops: 80000007 [#1] SMP ARM [ 208.688445] Modules linked in: snd_soc_wm8962 snd_soc_fsl_ssi snd_soc_imx_audmux imx_pcm_fiq evbug] [ 208.700662] CPU: 2 PID: 99 Comm: kworker/2:2 Not tainted 3.19.0-rc3-00069-g11c8f01-dirty #2 [ 208.709019] Hardware name: Freescale i.MX6 Quad/DualLite (Device Tree) [ 208.715575] Workqueue: events_power_efficient close_delayed_work [ 208.721605] task: be20ba80 ti: bdb40000 task.ti: bdb40000 [ 208.727012] PC is at 0x7f06541c [ 208.730164] LR is at snd_soc_dapm_set_bias_level+0x3c/0xc0 [ 208.735660] pc : [<7f06541c>] lr : [<805293e8>] psr: 200f0113 [ 208.735660] sp : bdb41da8 ip : bdb41dc8 fp : bdb41dc4 [ 208.747144] r10: bdb41e0c r9 : bc2ca9c4 r8 : bc2ca9ac [ 208.752375] r7 : bdb41e14 r6 : 00000001 r5 : bc2ca9cc r4 : bc2ca86c [ 208.758908] r3 : 7f06541c r2 : 00000001 r1 : bc2ca9cc r0 : bc2ca86c [ 208.765444] Flags: nzCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment kernel [ 208.772759] Control: 10c5387d Table: 4c05c04a DAC: 00000015 [ 208.778513] Process kworker/2:2 (pid: 99, stack limit = 0xbdb40238) [ 208.784787] Stack: (0xbdb41da8 to 0xbdb42000) [ 208.789154] 1da0: be20ba80 bc2ca9cc bc2ca9cc bc2ca86c bdb41ddc bdb41dc8 [ 208.797342] 1dc0: 8052ae88 805293b8 00000000 bc2ca9bc bdb41e4c bdb41de0 8052d194 8052ae4c [ 208.805529] 1de0: 80062798 8006259c bdb41e4c 00000002 bc2ca86c bc2ca9bc bdb41e0c bdb41e04 [ 208.813716] 1e00: bdb41e34 bdb41e04 bdb41e04 bdb41e0c bdb41e0c bdb41e14 bdb41e14 bdacd240 [ 208.821903] 1e20: 00000004 00000001 bb46b010 00000002 00000000 bc2ca8cc 00000000 00000001 [ 208.830089] 1e40: bdb41e74 bdb41e50 8052e5ac 8052ce20 00000000 bb46b5d4 bb46b01c bb46b010 [ 208.838276] 1e60: bdb41eb0 be7b6c00 bdb41e94 bdb41e78 8052fff8 8052e534 8052ffac be212680 [ 208.846462] 1e80: bb46b5d4 be7b3380 bdb41eec bdb41e98 8003ef74 8052ffb8 00000001 00000000 [ 208.854648] 1ea0: 8003ef08 8003f294 00000000 00000000 811cb670 80b2e650 00000000 808b0484 [ 208.862834] 1ec0: 806babe4 be212680 be7b33b0 809c1882 be212698 00000008 be7b3380 be7b3380 [ 208.871021] 1ee0: bdb41f24 bdb41ef0 8003f388 8003edec be214780 be212680 8003f228 00000000 [ 208.879207] 1f00: be214780 be212680 8003f228 00000000 00000000 00000000 bdb41fac bdb41f28 [ 208.887393] 1f20: 800446cc 8003f234 bdb41f44 00000000 80062798 be212680 00000000 00000000 [ 208.895579] 1f40: dead4ead ffffffff ffffffff 809c3f34 00000000 00000000 80832660 bdb41f5c [ 208.903764] 1f60: bdb41f5c 00000000 00000000 dead4ead ffffffff ffffffff 809c3f34 00000000 [ 208.911950] 1f80: 00000000 80832660 bdb41f88 bdb41f88 be214780 800445f0 00000000 00000000 [ 208.920137] 1fa0: 00000000 bdb41fb0 8000ed68 800445fc 00000000 00000000 00000000 00000000 [ 208.928323] 1fc0: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 [ 208.936509] 1fe0: 00000000 00000000 00000000 00000000 00000013 00000000 e89ffefb b6dafaea [ 208.944689] Backtrace: [ 208.947173] [<805293ac>] (snd_soc_dapm_set_bias_level) from [<8052ae88>] (dapm_pre_sequence_async+) [ 208.957004] r6:bc2ca86c r5:bc2ca9cc r4:bc2ca9cc r3:be20ba80 [ 208.962747] [<8052ae40>] (dapm_pre_sequence_async) from [<8052d194>] (dapm_power_widgets+0x380/0x7) [ 208.971970] r4:bc2ca9bc r3:00000000 [ 208.975593] [<8052ce14>] (dapm_power_widgets) from [<8052e5ac>] (snd_soc_dapm_stream_event+0x84/0x) [ 208.984816] r10:00000001 r9:00000000 r8:bc2ca8cc r7:00000000 r6:00000002 r5:bb46b010 [ 208.992736] r4:00000001 [ 208.995298] [<8052e528>] (snd_soc_dapm_stream_event) from [<8052fff8>] (close_delayed_work+0x4c/0x) [ 209.004522] r8:be7b6c00 r7:bdb41eb0 r6:bb46b010 r5:bb46b01c r4:bb46b5d4 r3:00000000 [ 209.012371] [<8052ffac>] (close_delayed_work) from [<8003ef74>] (process_one_work+0x194/0x40c) [ 209.020988] r6:be7b3380 r5:bb46b5d4 r4:be212680 r3:8052ffac [ 209.026725] [<8003ede0>] (process_one_work) from [<8003f388>] (worker_thread+0x160/0x48c) [ 209.034906] r10:be7b3380 r9:be7b3380 r8:00000008 r7:be212698 r6:809c1882 r5:be7b33b0 [ 209.042823] r4:be212680 [ 209.045384] [<8003f228>] (worker_thread) from [<800446cc>] (kthread+0xdc/0xf8) [ 209.052611] r10:00000000 r9:00000000 r8:00000000 r7:8003f228 r6:be212680 r5:be214780 [ 209.060530] r4:00000000 [ 209.063096] [<800445f0>] (kthread) from [<8000ed68>] (ret_from_fork+0x14/0x2c) [ 209.070322] r7:00000000 r6:00000000 r5:800445f0 r4:be214780 [ 209.076057] Code: bad PC value [ 209.079194] ---[ end trace dd15f2865dbe27f9 ]--- Seems it's because when playback stream is active soc_cleanup_card_resources() -> snd_card_free() Will trigger soc_pcm_close() which will queue close_delayed_work(), and then when close_delayed_work() is scheduled the necessary resource has already been released. I tried to set rtd->pmdown_time to 0 in soc_cleanup_card_resources() to avoid any new delayed_work be queued. BUT when unload audio drivers, I got another kernel crash ######################### [ 28.234740] Unable to handle kernel paging request at virtual address 6faaaf90 [ 28.242075] pgd = bbde0000 [ 28.244818] [6faaaf90] *pgd=00000000 [ 28.248511] Internal error: Oops: 5 [#1] SMP ARM [ 28.253155] Modules linked in: snd_soc_imx_wm8962(-) snd_soc_imx_audmux snd_soc_wm8962 snd_soc_fsl] [ 28.267055] CPU: 1 PID: 188 Comm: pulseaudio Not tainted 3.19.0-rc3-00069-g11c8f01-dirty #7 [ 28.275435] Hardware name: Freescale i.MX6 Quad/DualLite (Device Tree) [ 28.281991] task: bc3e1d40 ti: bbc1e000 task.ti: bbc1e000 [ 28.287431] PC is at dapm_seq_insert+0x34/0xf4 [ 28.291902] LR is at dapm_power_widgets+0x234/0x798 [ 28.296809] pc : [<8052ad74>] lr : [<8052d090>] psr: 600f0113 [ 28.296809] sp : bbc1fdd0 ip : 809bdd90 fp : bbc1fdec [ 28.308308] r10: bbc3bb90 r9 : bc0405c4 r8 : bc0405ac [ 28.313554] r7 : bbc1fe24 r6 : 809bdd90 r5 : bbc3b480 r4 : bbc3b134 [ 28.320093] r3 : bbc3b0c0 r2 : 00000000 r1 : bbc1fe1c r0 : bbc3bb40 [ 28.326628] Flags: nZCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment user [ 28.333771] Control: 10c5387d Table: 4bde004a DAC: 00000015 [ 28.339524] Process pulseaudio (pid: 188, stack limit = 0xbbc1e238) [ 28.345797] Stack: (0xbbc1fdd0 to 0xbbc20000) [ 28.350162] fdc0: 00000000 bbc3bb40 bbc3bba4 bbc1fe24 [ 28.358349] fde0: bbc1fe5c bbc1fdf0 8052d090 8052ad4c 80062798 8006259c bbc1fe5c 00000002 [ 28.366535] fe00: bc04046c bc0405bc bbc1fe1c bbc1fe14 bbc1fe44 bbc1fe14 bbc1fe14 bbc3b134 [ 28.374721] fe20: bbc3b134 bbc1fe24 bbc1fe24 bbc3bb40 00000004 00000001 bd916010 00000002 [ 28.382907] fe40: 00000000 bc0404cc 00000000 bd91601c bbc1fe84 bbc1fe60 8052e5f4 8052ce68 [ 28.391093] fe60: 00000000 bd916010 00000000 bc2c5a00 bbebb800 bdbe720c bbc1feac bbc1fe88 [ 28.399279] fe80: 805317cc 8052e57c bc2c5a00 bc2c5a00 bc3d9b80 bbdf7640 bc2c4ae8 00000008 [ 28.407465] fea0: bbc1fec4 bbc1feb0 8051b610 80531664 bc2c4a00 bc2c5a00 bbc1feec bbc1fec8 [ 28.415651] fec0: 8051b6a0 8051b5c0 8051b664 bbdf75c0 809bd508 bc3d9b80 bdba8f30 bdba8f30 [ 28.423837] fee0: bbc1ff0c bbc1fef0 8050eb2c 8051b670 bc3d9b80 bdba8f30 bd9b9850 bb8f75c0 [ 28.432023] ff00: bbc1ff4c bbc1ff10 800ef5bc 8050ea50 00000000 00000000 80042d68 bc3d9b88 [ 28.440208] ff20: bc3d9b80 bc3e20f0 00000000 809c9f14 bc3e1d40 00000000 bbc1e000 00000000 [ 28.448393] ff40: bbc1ff5c bbc1ff50 800ef760 800ef538 bbc1ff84 bbc1ff60 80042ec0 800ef75c [ 28.456579] ff60: bc3d9b80 8000ee64 bbc1ffb0 00000000 00000006 8000ee64 bbc1ffac bbc1ff88 [ 28.464765] ff80: 80011884 80042e14 8000ecac 00000004 00000000 0116aa50 0116ab58 0118e100 [ 28.472950] ffa0: 00000000 bbc1ffb0 8000ecf8 800117dc 00000000 76fcf084 76fcf4c0 725fcc94 [ 28.481136] ffc0: 0116aa50 0116ab58 0118e100 00000006 01118168 4ce839e4 00000000 00000000 [ 28.489322] ffe0: 00000000 7e957978 4e1e6d24 4e1e6d34 800f0010 00000016 4eff6821 4eff6c21 [ 28.497501] Backtrace: [ 28.499980] [<8052ad40>] (dapm_seq_insert) from [<8052d090>] (dapm_power_widgets+0x234/0x798) [ 28.508509] r7:bbc1fe24 r6:bbc3bba4 r5:bbc3bb40 r4:00000000 [ 28.514251] [<8052ce5c>] (dapm_power_widgets) from [<8052e5f4>] (snd_soc_dapm_stream_event+0x84/0x) [ 28.523473] r10:bd91601c r9:00000000 r8:bc0404cc r7:00000000 r6:00000002 r5:bd916010 [ 28.531393] r4:00000001 [ 28.533959] [<8052e570>] (snd_soc_dapm_stream_event) from [<805317cc>] (soc_pcm_close+0x174/0x28c) [ 28.542920] r8:bdbe720c r7:bbebb800 r6:bc2c5a00 r5:00000000 r4:bd916010 r3:00000000 [ 28.550764] [<80531658>] (soc_pcm_close) from [<8051b610>] (snd_pcm_release_substream+0x5c/0xb0) [ 28.559552] r10:00000008 r8:bc2c4ae8 r7:bbdf7640 r6:bc3d9b80 r5:bc2c5a00 r4:bc2c5a00 [ 28.567479] [<8051b5b4>] (snd_pcm_release_substream) from [<8051b6a0>] (snd_pcm_release+0x3c/0x88) [ 28.576440] r5:bc2c5a00 r4:bc2c4a00 [ 28.580062] [<8051b664>] (snd_pcm_release) from [<8050eb2c>] (snd_disconnect_release+0xe8/0xf8) [ 28.588764] r8:bdba8f30 r7:bdba8f30 r6:bc3d9b80 r5:809bd508 r4:bbdf75c0 r3:8051b664 [ 28.596604] [<8050ea44>] (snd_disconnect_release) from [<800ef5bc>] (__fput+0x90/0x1d4) [ 28.604611] r7:bb8f75c0 r6:bd9b9850 r5:bdba8f30 r4:bc3d9b80 [ 28.610345] [<800ef52c>] (__fput) from [<800ef760>] (____fput+0x10/0x14) [ 28.617050] r10:00000000 r9:bbc1e000 r8:00000000 r7:bc3e1d40 r6:809c9f14 r5:00000000 [ 28.624968] r4:bc3e20f0 [ 28.627535] [<800ef750>] (____fput) from [<80042ec0>] (task_work_run+0xb8/0xe8) [ 28.634865] [<80042e08>] (task_work_run) from [<80011884>] (do_work_pending+0xb4/0xd4) [ 28.642786] r8:8000ee64 r7:00000006 r6:00000000 r5:bbc1ffb0 r4:8000ee64 r3:bc3d9b80 [ 28.650624] [<800117d0>] (do_work_pending) from [<8000ecf8>] (work_pending+0xc/0x20) [ 28.658371] r6:0118e100 r5:0116ab58 r4:0116aa50 [ 28.663047] Code: e5905000 e3520000 e24c6070 01a0600c (e7965105) [ 28.669220] ---[ end trace 69b76cbdebd80b05 ]--- >From the backtrace, seems it's because when soc_pcm_close() is called, it will power widget due to stream event, but at that time, all widgets have already been freed by soc_remove_dai_links() Currently I am out of idea how to fix this issue, As I think this is a generic issue not i.MX platform specific, Someone in community may has already find this issue, Could anyone give some inputs/suggestion to fix this issue? Regards & Thanks, Jiada

6 19

[alsa-devel] ASoC Machine driver
by Ghadiya, Anil 15 Jan '15

15 Jan '15

Hi All, I am new to ALSA architecture and want to understand the flow of the driver. So I took one of the existing codec driver "adau1701" and build it for the my imx6q-sabrelite development board. I am able to build and insert it on the board. Now I want to write ASoC machine driver for the same. I am hoping that I do not need to write ASoC platform driver as my codec is not using any I2S, DMA, S/PDIF or any SOC specific driver. I have seen the machine driver for other platform. I can see that sound card is getting registered in the probe function of the machine driver, but . I am not able to understand the flow here. Could you please help me understand the flow of execution like........ when probe function of machine driver gets called? When will the probe function od codec driver gets executed? and so on.... Thanks in Advance, Anil Patel

1 0

[alsa-devel] [PATCH] ALSA: hda - Select INPUT for Realtek HD-audio codec
by Takashi Iwai 15 Jan '15

15 Jan '15

The commit commit [33f4acd3b214: ALSA: hda - Enable mic mute hotkey and LEDs for an HP machine] introduced a quirk for a HP machine involving with the input event handling. Although the relevant code is protected via IS_ENABLED(CONFIG_INPUT), this doesn't suffice when the audio driver is built in while the input is module. As an easy workaround, this patch forcibly selects CONFIG_INPUT in Kconfig. This shouldn't be a practical problem since CONFIG_INPUT is almost mandatory for all systems. Also, this allows to remove the ugly ifdefs in the code. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/Kconfig | 1 + sound/pci/hda/patch_realtek.c | 6 ------ 2 files changed, 1 insertion(+), 6 deletions(-) diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index ebf4c2fb99df..7f0f2c5a4e97 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -107,6 +107,7 @@ config SND_HDA_PATCH_LOADER config SND_HDA_CODEC_REALTEK tristate "Build Realtek HD-audio codec support" select SND_HDA_GENERIC + select INPUT help Say Y or M here to include Realtek HD-audio codec support in snd-hda-intel driver, such as ALC880. diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a50e15e166c6..1720f8a457d1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -121,9 +121,7 @@ struct alc_spec { hda_nid_t pll_nid; unsigned int pll_coef_idx, pll_coef_bit; unsigned int coef0; -#if IS_ENABLED(CONFIG_INPUT) struct input_dev *kb_dev; -#endif }; /* @@ -3476,7 +3474,6 @@ static void alc280_fixup_hp_gpio4(struct hda_codec *codec, } } -#if IS_ENABLED(CONFIG_INPUT) static void gpio2_mic_hotkey_event(struct hda_codec *codec, struct hda_jack_callback *event) { @@ -3489,12 +3486,10 @@ static void gpio2_mic_hotkey_event(struct hda_codec *codec, input_report_key(spec->kb_dev, KEY_MICMUTE, 0); input_sync(spec->kb_dev); } -#endif static void alc280_fixup_hp_gpio2_mic_hotkey(struct hda_codec *codec, const struct hda_fixup *fix, int action) { -#if IS_ENABLED(CONFIG_INPUT) /* GPIO1 = set according to SKU external amp GPIO2 = mic mute hotkey GPIO3 = mute LED @@ -3550,7 +3545,6 @@ static void alc280_fixup_hp_gpio2_mic_hotkey(struct hda_codec *codec, input_unregister_device(spec->kb_dev); spec->kb_dev = NULL; } -#endif } static void alc269_fixup_hp_line1_mic1_led(struct hda_codec *codec, -- 2.2.1

2 1

[alsa-devel] [PATCH] ASoC: sirf: fix platform_no_drv_owner.cocci warnings
by kbuild test robot 15 Jan '15

15 Jan '15

sound/soc/sirf/sirf-atlas7-cs42xx8.c:192:3-8: No need to set .owner here. The core will do it. Remove .owner field if calls are used which set it automatically Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci CC: Yibo Cai <yibo.cai(a)csr.com> Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> --- sirf-atlas7-cs42xx8.c | 1 - 1 file changed, 1 deletion(-) --- a/sound/soc/sirf/sirf-atlas7-cs42xx8.c +++ b/sound/soc/sirf/sirf-atlas7-cs42xx8.c @@ -189,7 +189,6 @@ MODULE_DEVICE_TABLE(of, sirf_hdmi_card_o static struct platform_driver sirf_hdmi_card_driver = { .driver = { .name = "sirf-hdmi-card", - .owner = THIS_MODULE, .pm = &snd_soc_pm_ops, .of_match_table = sirf_hdmi_card_of_match, },

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[alsa-devel] [PATCH - UCM 1/1] ucm: add binary configure file parse
by han.lu＠intel.com 15 Jan '15

15 Jan '15

From: "Lu, Han" <han.lu(a)intel.com> with cset command, UCM set kcontrol parameters directly: cset "name='<KCONTROL_NAME>' 1<,2,3,...>" This patch enables UCM to set kcontrol with parameters from configure file: cset-bin-file "name='<KCONTROL_NAME>' <path/to/file>" where "cset-bin-file" is a newly added keyword alongside of "cset", to indicate cset with binary data in file. The binary data is parameter only for audio DSPs, and it's just passed by UCM/ALSA as raw data. The data type of the parameter element must be byte, and the maxim number of parameter elements is 512 (the maxim value that struct snd_ctl_elem_value can hold). Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/src/ucm/main.c b/src/ucm/main.c index 37ae4c8..d2b1e15 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -160,11 +160,45 @@ static int open_ctl(snd_use_case_mgr_t *uc_mgr, return 0; } +static int binary_file_parse(snd_ctl_elem_value_t *dst, + const char *filepath) +{ + int err = 0; + FILE *in; + long len; + char *res; + unsigned int idx; + + in = fopen(filepath, "r"); + if (!in) { + err = -errno; + goto __fail; + } + fseek(in, 0L, SEEK_END); + len = ftell(in); + rewind(in); + if (len > 512) + len = 512; + res = calloc(1, (size_t)len); + if (res == NULL) { + err = -ENOMEM; + goto __fail_nomem; + } + fread(res, (size_t)len, 1, in); + for (idx = 0; idx < len; idx++) + snd_ctl_elem_value_set_byte(dst, idx, *(res + idx)); + free(res); + __fail_nomem: + fclose(in); + __fail: + return err; +} + extern int __snd_ctl_ascii_elem_id_parse(snd_ctl_elem_id_t *dst, const char *str, const char **ret_ptr); -static int execute_cset(snd_ctl_t *ctl, const char *cset) +static int execute_cset(snd_ctl_t *ctl, const char *cset, int isbin) { const char *pos; int err; @@ -194,7 +228,10 @@ static int execute_cset(snd_ctl_t *ctl, const char *cset) err = snd_ctl_elem_info(ctl, info); if (err < 0) goto __fail; - err = snd_ctl_ascii_value_parse(ctl, value, info, pos); + if (isbin) + err = binary_file_parse(value, pos); + else + err = snd_ctl_ascii_value_parse(ctl, value, info, pos); if (err < 0) goto __fail; err = snd_ctl_elem_write(ctl, value); @@ -239,6 +276,7 @@ static int execute_sequence(snd_use_case_mgr_t *uc_mgr, goto __fail_nomem; break; case SEQUENCE_ELEMENT_TYPE_CSET: + case SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE: if (cdev == NULL) { const char *cdev1 = NULL, *cdev2 = NULL; err = get_value3(&cdev1, "PlaybackCTL", @@ -274,7 +312,7 @@ static int execute_sequence(snd_use_case_mgr_t *uc_mgr, goto __fail; } } - err = execute_cset(ctl, s->data.cset); + err = execute_cset(ctl, s->data.cset, s->isbin); if (err < 0) { uc_error("unable to execute cset '%s'\n", s->data.cset); goto __fail; diff --git a/src/ucm/parser.c b/src/ucm/parser.c index d7517f6..fb5a033 100644 --- a/src/ucm/parser.c +++ b/src/ucm/parser.c @@ -306,6 +306,17 @@ static int parse_sequence(snd_use_case_mgr_t *uc_mgr ATTRIBUTE_UNUSED, continue; } + if (strcmp(cmd, "cset-bin-file") == 0) { + curr->type = SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE; + curr->isbin = 1; + err = parse_string(n, &curr->data.cset); + if (err < 0) { + uc_error("error: cset-bin-file requires a string!"); + return err; + } + continue; + } + if (strcmp(cmd, "usleep") == 0) { curr->type = SEQUENCE_ELEMENT_TYPE_SLEEP; err = snd_config_get_integer(n, &curr->data.sleep); diff --git a/src/ucm/ucm_local.h b/src/ucm/ucm_local.h index 87f14a2..d0a609b 100644 --- a/src/ucm/ucm_local.h +++ b/src/ucm/ucm_local.h @@ -47,6 +47,7 @@ #define SEQUENCE_ELEMENT_TYPE_CSET 2 #define SEQUENCE_ELEMENT_TYPE_SLEEP 3 #define SEQUENCE_ELEMENT_TYPE_EXEC 4 +#define SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE 5 struct ucm_value { struct list_head list; @@ -63,6 +64,7 @@ struct sequence_element { char *cset; char *exec; } data; + int isbin; /* Indicate cset to handle binary or ascii array */ }; /* -- 1.9.1

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[alsa-devel] Configuring slave output for ALC889
by Dheeraj CVR 15 Jan '15

15 Jan '15

Hi, I am using an Intel Desktop Board DX58SO. I am using S/PDIF for HDMI Audio and there is no sound output via HDMI. I have checked with alsamixer and the volumes for the S/PDIF are stuck at 0 and I can't increase them. By looking at the commit history, I have found this commit which enables 0x10 as a slave output for 0x6 on ALC889 for the exact same board that I am using. http://mailman.alsa-project.org/pipermail/alsa-devel/2009-April/015901.html However, it appears that with the commit "ALSA: hda/realtek - Drop ALC882 desktop model quirks", the older workaround has been removed, which caused a regression. I have checked the output of alsainfo and it seems that 0x06 is still being used instead of 0x010. Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Control: name="SPDIF Phantom Jack", index=1, device=0 Pincap 0x00000010: OUT Pin Default 0x99430130: [Fixed] SPDIF Out at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x06 I was planning to edit the "patch_alc882" function to configure the "spec->gen.slave_dig_outs" and set the values to "{0x10, 0, 0}". However, I was not sure if this was the right way of doing. Please advice. I am running Arch-Linux with 3.17 kernel. -- Regards, Dheeraj CVR.

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[alsa-devel] [PATCH] ASoC: Intel: initial stream_hw_id to invalid value
by Jie Yang 14 Jan '15

14 Jan '15

The stream_hw_id for System stream is 0x0, if we use initial stream_hw_id value 0, it may return wrong(not committed) stream when calling function get_stream_by_id() with stream_id=0. Here initial stream_hw_id to invalid value to fix this issue. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-haswell-ipc.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 5bf1404..eeef9a5 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -94,6 +94,8 @@ /* Mailbox */ #define IPC_MAX_MAILBOX_BYTES 256 +#define INVALID_STREAM_HW_ID 0xffffffff + /* Global Message - Types and Replies */ enum ipc_glb_type { IPC_GLB_GET_FW_VERSION = 0, /* Retrieves firmware version */ @@ -1208,6 +1210,7 @@ struct sst_hsw_stream *sst_hsw_stream_new(struct sst_hsw *hsw, int id, return NULL; spin_lock_irqsave(&sst->spinlock, flags); + stream->reply.stream_hw_id = INVALID_STREAM_HW_ID; list_add(&stream->node, &hsw->stream_list); stream->notify_position = notify_position; stream->pdata = data; -- 1.9.1

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