November 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: soc-pcm: skip dpcm path checking with incapable/unready FE
by Qiao Zhou 24 Nov '14

24 Nov '14

Skip dpcm path checking for playback or capture, if corresponding FE doesn't support playback or capture, or currently is not ready. It can reduce the unnecessary cost to search connected widgets. Signed-off-by: Qiao Zhou <zhouqiao(a)marvell.com> --- sound/soc/soc-pcm.c | 16 ++++++++++++++-- 1 file changed, 14 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 002311a..3b068d4 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2248,7 +2248,13 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card) fe->dai_link->name); /* skip if FE doesn't have playback capability */ - if (!fe->cpu_dai->driver->playback.channels_min) + if (!fe->cpu_dai->driver->playback.channels_min + || !fe->codec_dai->driver->playback.channels_min) + goto capture; + + /* skip if FE doesn't do playback */ + if (!fe->cpu_dai->playback_active + || !fe->codec_dai->playback_active) goto capture; paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list); @@ -2278,7 +2284,13 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card) dpcm_path_put(&list); capture: /* skip if FE doesn't have capture capability */ - if (!fe->cpu_dai->driver->capture.channels_min) + if (!fe->cpu_dai->driver->capture.channels_min + || !fe->codec_dai->driver->capture.channels_min) + continue; + + /* skip if FE doesn't do capture */ + if (!fe->cpu_dai->capture_active + || !fe->codec_dai->capture_active) continue; paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list); -- 1.7.9.5

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[alsa-devel] Debian Bug#770122: patch as workaround/solution
by Elimar Riesebieter 23 Nov '14

23 Nov '14

Hi, Iam forwarding this Debian bug to alsa-devel due to a not maintained BugTracker at alsa project. Could one please have a look at it? Thanks Elimar -- Alles was viel bedacht wird ist bedenklich!;-) Friedrich Nietzsche

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[alsa-devel] [PATCH 0/10] fix error return code
by Julia Lawall 23 Nov '14

23 Nov '14

These patches fix cases where the return code appears to be unintentially nonnegative. The complete semantic match that finds the problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @ok exists@ identifier f,ret,i; expression e; constant c; @@ f(...) { <+... ( return -c@i; | ret = -c@i; ... when != ret = e return ret; | if (ret < 0) { ... return ret; } ) ...+> } @r exists@ identifier ret,l,ok.f; expression e1,e2,e3; statement S; position p1,p2,p3; @@ f(...) { ... when any ( if@p1 ($ret < 0\|ret != 0$) { ... return ret; } | ret@p1 = 0 ) ... when != ret = e1 when != &ret ( if (<+... ret = e3 ...+>) S | if (<+... &ret ...+>) S | if@p2(...) { ... when != ret = e2 when forall return@p3 ret; } ) ... when any } @bad exists@ position r.p1,r.p2; statement S1,S2; identifier r.ret; expression e1; @@ ( if@p1 ($ret < 0\|ret != 0$) S1 | ret@p1 = 0 ) ... when any ret = e1 ... when any if@p2(...) S2 @bad2@ position r.p1,r.p2; identifier r.ret; expression e1; statement S2; @@ ret@p1 = 0 ... when != if (...) { ... ret = e1 ... return ret; } when any if@p2(...) S2 @script:python depends on !bad && !bad2@ p1 << r.p1; p2 << r.p2; p3 << r.p3; @@ cocci.print_main("",p1) cocci.print_secs("",p2) cocci.print_secs("",p3) // </smpl>

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[alsa-devel] Debian Bug#770122: patch as workaround/solution
by Elimar Riesebieter 23 Nov '14

23 Nov '14

Hi, I am forwarding this Debian bug [0] to alsa-devel due to a not maintained BugTracker at alsa project. Could one please have a look at it? [0] http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=770122 Thanks Elimar -- Alles was viel bedacht wird ist bedenklich!;-) Friedrich Nietzsche

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[alsa-devel] [PATCH 1/2] amixer: Make "dB" case-insensitive in set commands
by Takashi Iwai 23 Nov '14

23 Nov '14

We don't have to be necessarily too strict about case-sensitivity of "dB" suffix used in set commands. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- amixer/amixer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/amixer/amixer.c b/amixer/amixer.c index cf82892bfa2f..6a2fdb96c62f 100644 --- a/amixer/amixer.c +++ b/amixer/amixer.c @@ -346,7 +346,7 @@ static int set_volume_simple(snd_mixer_elem_t *elem, if (*p == '%') { percent = 1; p++; - } else if (p[0] == 'd' && p[1] == 'B') { + } else if (toupper(p[0]) == 'D' && toupper(p[1]) == 'B') { vol_type = VOL_DB; p += 2; scale = 100; -- 2.1.3

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[alsa-devel] [PATCH 00/10] ALSA: asihpi: updates
by eliot＠blennerhassett.gen.nz 23 Nov '14

23 Nov '14

From: Eliot Blennerhassett <eliot(a)blennerhassett.gen.nz> Major change - Add support for interrupt driven 'low latency' mode PCM streams on some AudioScience adapters. Various other minor updates from AudioScience Eliot Blennerhassett (10): ALSA: asihpi: Minor string and dead code cleanup ALSA: asihpi: New I/O types - AVB & BLUlink, DAB Rf receiver ALSA: asihpi: Logging format improvements ALSA: asihpi: Use CONFIG_64BIT directly ALSA: asihpi: Refactor control cache code. ALSA: asihpi: Add support for stream interrupt. ALSA: asihpi: Turn off msg/resp logging after DSP has crashed. ALSA: asihpi: Use standard printk helpers ALSA: asihpi: don't fail probe if adapter mode read fails ALSA: asihpi: used parts of message/response are zeroed before use sound/pci/asihpi/asihpi.c | 326 +++++++++++++++++++++++++++------------- sound/pci/asihpi/hpi.h | 16 +- sound/pci/asihpi/hpi6205.c | 43 +++++- sound/pci/asihpi/hpi_internal.h | 20 ++- sound/pci/asihpi/hpicmn.c | 107 +++++++------ sound/pci/asihpi/hpicmn.h | 19 ++- sound/pci/asihpi/hpimsginit.c | 30 ++-- sound/pci/asihpi/hpimsgx.c | 15 +- sound/pci/asihpi/hpioctl.c | 123 +++++++++++++-- sound/pci/asihpi/hpios.h | 8 +- 10 files changed, 508 insertions(+), 199 deletions(-) -- 1.9.1

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[alsa-devel] randconfig errors in ts3a227e.c
by devendra.aaru 23 Nov '14

23 Nov '14

Hello All, While testing the build, i have got the following errors: sound/soc/codecs/ts3a227e.c: In function ‘ts3a227e_enable_jack_detect’: sound/soc/codecs/ts3a227e.c:223:2: error: implicit declaration of function ‘snd_jack_set_key’ [-Werror=implicit-function-declaration] snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); ^ cc1: some warnings being treated as errors make[3]: *** [sound/soc/codecs/ts3a227e.o] Error 1 The CONFIG_SND_JACK is not enabled in the .config and the #else portion of include/sound/jack.h doesn't contain snd_jack_set_key stub. This resulting in the build error. I am unsure if adding stub in include/sound/jack.h or adding a Kconfig knob to deselect the ts3a227e.c fixes the issue ? full randconfig attached. (sorry about the 55k file size)

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[alsa-devel] [patch] sound: oss: uart401: remove unneeded NULL check
by Dan Carpenter 22 Nov '14

22 Nov '14

"devc" can't be NULL here so there is no need to check. Also I removed the "devc = NULL" assignment because devc is stored on stack so it's a no-op. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/oss/uart401.c b/sound/oss/uart401.c index 279bc56..dae4d43 100644 --- a/sound/oss/uart401.c +++ b/sound/oss/uart401.c @@ -412,13 +412,10 @@ void unload_uart401(struct address_info *hw_config) if (!devc->share_irq) free_irq(devc->irq, devc); - if (devc) - { - kfree(midi_devs[devc->my_dev]->converter); - kfree(midi_devs[devc->my_dev]); - kfree(devc); - devc = NULL; - } + kfree(midi_devs[devc->my_dev]->converter); + kfree(midi_devs[devc->my_dev]); + kfree(devc); + /* This kills midi_devs[x] */ sound_unload_mididev(hw_config->slots[4]); }

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[alsa-devel] Headset jack and intel based Macmini6,2
by Marco 22 Nov '14

22 Nov '14

/cc David Henningsson that has done something like that for some Dell HW and might have some suggestions Hello, even if poorly advertised, recent mac.* have an headset jack (amplified) along with the non amplified line jack. For example, for MacMini, from: apple.com/mac-mini/specs/ > Support for Apple iPhone headset with microphone Where iPhone headset means a simple headset not preamplified, really. I own a MacMini6,2 (the october 2012 one) and with Mac OSX it works... under linux... well... it is a simple out jack. This jeopardize the ability to use any video conference software (who doesn't today?) on linux if you don't own a USB audio device (or you don't want that extra piece of relatively large hw) The card is a HDA based Cirrus Logic CS4206/4207 and the two specific models do not work (imac27 and mbp55) while auto works (except the headset of course :) I tried to change the pinconfig but on the intel HDA documentation it is not really clear how the code for an headset jack (in/out) would look like. I saw that for card ALC269/270/275/276/28x/29x there are models that support the headset codes but... well.. they don't really seem "standard" intel codes I tried to look if it was possible to extract these codes from OSX but I haven't found anything interesting (but I might have missed something) Any suggestion dear esteemed Alsa developers? :) Any help appreciated, Thanks! Marco

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[alsa-devel] [PATCH] ASoC: Intel: Add Cherrytrail & Braswell machine driver cht_bsw_rt5672
by mengdong.lin＠intel.com 21 Nov '14

21 Nov '14

From: Mengdong Lin <mengdong.lin(a)intel.com> Add machine driver for two Intel Cherryview-based platforms, Cherrytrail and Braswell, with RT5672 codec. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 2a3af88..7479ce0 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -16,6 +16,9 @@ config SND_SST_MFLD_PLATFORM config SND_SST_IPC tristate +config SND_SST_MACHINE + tristate + config SND_SOC_INTEL_SST tristate "ASoC support for Intel(R) Smart Sound Technology" select SND_SOC_INTEL_SST_ACPI if ACPI @@ -76,3 +79,20 @@ config SND_SOC_INTEL_BROADWELL_MACH Ultrabook platforms. Say Y if you have such a device If unsure select "N". + +config SND_SOC_INTEL_CHT_BSW_RT5672_MACH + tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5672 codec" + depends on X86_INTEL_LPSS + select SND_SOC_RT5670 + select SND_SST_MFLD_PLATFORM + select SND_SOC_INTEL_SST + select SND_SST_IPC + select SND_SST_MACHINE + default n + + help + This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell + platforms with RT5672 audio codec. + Say Y if you have such a device + If unsure select "N". + diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 9ab43be..4069d3f 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -31,6 +31,7 @@ obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o +obj-$(CONFIG_SND_SST_MACHINE) += board/ # DSP driver obj-$(CONFIG_SND_SST_IPC) += sst/ diff --git a/sound/soc/intel/board/Makefile b/sound/soc/intel/board/Makefile new file mode 100644 index 0000000..9ecc227 --- /dev/null +++ b/sound/soc/intel/board/Makefile @@ -0,0 +1,2 @@ +snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o +obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o diff --git a/sound/soc/intel/board/cht_bsw_rt5672.c b/sound/soc/intel/board/cht_bsw_rt5672.c new file mode 100644 index 0000000..dffd8b1 --- /dev/null +++ b/sound/soc/intel/board/cht_bsw_rt5672.c @@ -0,0 +1,286 @@ +/* + * cht_bsw_rt5672.c - ASoc Machine driver for Intel Cherryview-based platforms + * Cherrytrail and Braswell, with RT5672 codec. + * + * Copyright (C) 2014 Intel Corp + * Author: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> + * Mengdong Lin <mengdong.lin(a)intel.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "../../codecs/rt5670.h" +#include "../sst-atom-controls.h" + +#define CHT_PLAT_CLK_3_HZ 19200000 + +static const struct snd_soc_dapm_widget cht_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static const struct snd_soc_dapm_route cht_audio_map[] = { + {"IN1P", NULL, "Headset Mic"}, + {"IN1N", NULL, "Headset Mic"}, + {"DMIC L1", NULL, "Int Mic"}, + {"DMIC R1", NULL, "Int Mic"}, + {"Headphone", NULL, "HPOL"}, + {"Headphone", NULL, "HPOR"}, + {"Ext Spk", NULL, "SPOLP"}, + {"Ext Spk", NULL, "SPOLN"}, + {"Ext Spk", NULL, "SPORP"}, + {"Ext Spk", NULL, "SPORN"}, + {"AIF1 Playback", NULL, "ssp2 Tx"}, + {"ssp2 Tx", NULL, "codec_out0"}, + {"ssp2 Tx", NULL, "codec_out1"}, + {"codec_in0", NULL, "ssp2 Rx"}, + {"codec_in1", NULL, "ssp2 Rx"}, + {"ssp2 Rx", NULL, "AIF1 Capture"}, +}; + +static const struct snd_kcontrol_new cht_mc_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Int Mic"), + SOC_DAPM_PIN_SWITCH("Ext Spk"), +}; + +static int cht_aif1_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + unsigned int fmt; + + if (strncmp(codec_dai->name, "rt5670-aif1", 11)) + return 0; + + /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, + SNDRV_PCM_FORMAT_GSM); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec TDM slot %d\n", ret); + return ret; + } + + /* TDM slave Mode */ + fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF + | SND_SOC_DAIFMT_CBS_CFS; + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec DAI fmt %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5670_PLL1_S_MCLK, + CHT_PLAT_CLK_3_HZ, params_rate(params) * 512); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec pll: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_PLL1, + params_rate(params) * 512, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec sysclk: %d\n", ret); + return ret; + } + return 0; +} + +static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* The DSP will covert the FE rate to 48k, stereo, 24bits */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP2 to 24-bit */ + snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT - + SNDRV_PCM_HW_PARAM_FIRST_MASK], + SNDRV_PCM_FORMAT_S24_LE); + return 0; +} + +static int cht_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + case SND_SOC_BIAS_OFF: + break; + default: + dev_err(card->dev, "Invalid bias level=%d\n", level); + return -EINVAL; + } + card->dapm.bias_level = level; + return 0; +} + +static int cht_init(struct snd_soc_pcm_runtime *runtime) +{ + int ret; + struct snd_soc_card *card = runtime->card; + + /* Set card bias level */ + cht_set_bias_level(card, &card->dapm, SND_SOC_BIAS_OFF); + card->dapm.idle_bias_off = true; + + ret = snd_soc_dapm_sync(&card->dapm); + if (ret) { + dev_err(card->dev, "unable to sync dapm\n"); + return ret; + } + return ret; +} + +static unsigned int rates_48000[] = { + 48000, +}; + +static struct snd_pcm_hw_constraint_list constraints_48000 = { + .count = ARRAY_SIZE(rates_48000), + .list = rates_48000, +}; + +static int cht_aif1_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_48000); +} + +static struct snd_soc_ops cht_aif1_ops = { + .startup = cht_aif1_startup, +}; + +static struct snd_soc_ops cht_be_ssp2_ops = { + .hw_params = cht_aif1_hw_params, +}; + +static struct snd_soc_dai_link cht_dailink[] = { + /* Front End DAI links */ + [MERR_DPCM_AUDIO] = { + .name = "Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "media-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + .init = cht_init, + .ignore_suspend = 1, + .dynamic = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &cht_aif1_ops, + }, + [MERR_DPCM_COMPR] = { + .name = "Compressed Port", + .stream_name = "Compress", + .cpu_dai_name = "compress-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + }, + + /* Back End DAI links */ + { + /* SSP2 - Codec */ + .name = "SSP2-Codec", + .be_id = 1, + .cpu_dai_name = "ssp2-port", + .platform_name = "sst-mfld-platform", + .no_pcm = 1, + .codec_dai_name = "rt5670-aif1", + .codec_name = "i2c-10EC5670:00", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .be_hw_params_fixup = cht_codec_fixup, + .ignore_suspend = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &cht_be_ssp2_ops, + }, +}; + +/* SoC card */ +static struct snd_soc_card snd_soc_card_cht = { + .name = "cherrytrailcraudio", + .dai_link = cht_dailink, + .num_links = ARRAY_SIZE(cht_dailink), + .set_bias_level = cht_set_bias_level, + .dapm_widgets = cht_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets), + .dapm_routes = cht_audio_map, + .num_dapm_routes = ARRAY_SIZE(cht_audio_map), + .controls = cht_mc_controls, + .num_controls = ARRAY_SIZE(cht_mc_controls), +}; + +static int snd_cht_mc_probe(struct platform_device *pdev) +{ + int ret_val = 0; + + /* register the soc card */ + snd_soc_card_cht.dev = &pdev->dev; + ret_val = snd_soc_register_card(&snd_soc_card_cht); + if (ret_val) { + dev_err(&pdev->dev, + "snd_soc_register_card failed %d\n", ret_val); + return ret_val; + } + platform_set_drvdata(pdev, &snd_soc_card_cht); + return ret_val; +} + +static int snd_cht_mc_remove(struct platform_device *pdev) +{ + struct snd_soc_card *soc_card = platform_get_drvdata(pdev); + + snd_soc_card_set_drvdata(soc_card, NULL); + snd_soc_unregister_card(soc_card); + platform_set_drvdata(pdev, NULL); + return 0; +} + +static struct platform_driver snd_cht_mc_driver = { + .driver = { + .owner = THIS_MODULE, + .name = "cht-bsw-rt5672", + .pm = &snd_soc_pm_ops, + }, + .probe = snd_cht_mc_probe, + .remove = snd_cht_mc_remove, +}; + +module_platform_driver(snd_cht_mc_driver); + +MODULE_DESCRIPTION("ASoC Intel(R) Baytrail CR Machine driver"); +MODULE_AUTHOR("Subhransu S. Prusty, Mengdong Lin"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:cht-bsw-rt5672"); -- 1.9.1

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