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

[alsa-devel] [Intel HDA][ALC3226] Too much white noise from microphone
by Shawn Starr 07 Feb '15

07 Feb '15

Hello ALSA devs, I spoke to patrakov on Freenode from the #Alsa IRC channel and they mentioned that the noise background I am experiencing is not normal. I tried two different headsets, both headsets have two separate jacks, the same result. I will test any patches/quirks for kernel as needed. Here is the alsa-info: http://www.alsa-project.org/db/?f=29d3c45e2d66c58be03a10bd0186067a9d735a11 Here is a 10 second recording: http://turl.ca/zrhbcs Here is the amixer information: Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 87 Mono: Playback 87 [100%] [0.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 87 Mono: Front Left: Playback 87 [100%] [0.00dB] [on] Front Right: Playback 87 [100%] [0.00dB] [on] Simple mixer control 'Speaker',0 Capabilities: pswitch Playback channels: Front Left - Front Right Mono: Front Left: Playback [off] Front Right: Playback [off] Simple mixer control 'Speaker+LO',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 87 Mono: Front Left: Playback 65 [75%] [-16.50dB] Front Right: Playback 65 [75%] [-16.50dB] Simple mixer control 'PCM',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.00dB] Front Right: Playback 255 [100%] [0.00dB] Simple mixer control 'Front Mic',0 Capabilities: pvolume pswitch cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Mono Limits: Playback 0 - 31 Mono: Capture [on] Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Front Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 2 [67%] [20.00dB] Front Right: 2 [67%] [20.00dB] Simple mixer control 'Line Out',0 Capabilities: pswitch Playback channels: Front Left - Front Right Mono: Front Left: Playback [off] Front Right: Playback [off] Simple mixer control 'Beep',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 63 Front Left: Capture 63 [100%] [30.00dB] [on] Front Right: Capture 63 [100%] [30.00dB] [on] Simple mixer control 'Auto-Mute Mode',0 Capabilities: enum Items: 'Disabled' 'Speaker Only' 'Line Out+Speaker' Item0: 'Line Out+Speaker' Simple mixer control 'Dock Mic',0 Capabilities: pvolume pswitch cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Mono Limits: Playback 0 - 31 Mono: Capture [off] Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Dock Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right

3 5

[alsa-devel] [PATCH v2 0/3] ASoC: add audio card creation from graph of ports in DT
by Jean-Francois Moine 07 Feb '15

07 Feb '15

This patchset adds a way to build audio cards from the description of a graph of ports in a devicetree. v2: - move the soc-core and dt-card from the patchset "ASoC: tda998x: add a codec to the HDMI transmitter" - add DT documentation (Mark Brown) Jean-Francois Moine (3): ASoC: core: export snd_soc_get_dai_name Documentation: of: Document audio graph bindings ASoC: add generic dt-card support .../devicetree/bindings/sound/audio-interfaces.txt | 101 ++++++++ include/sound/soc.h | 2 + sound/soc/generic/Kconfig | 2 + sound/soc/generic/Makefile | 2 + sound/soc/generic/dt-card.c | 275 +++++++++++++++++++++ sound/soc/soc-core.c | 5 +- 6 files changed, 385 insertions(+), 2 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-interfaces.txt create mode 100644 sound/soc/generic/dt-card.c -- 2.1.4

3 20

[alsa-devel] [PATCH v3 0/15] Those patches is used for dw_hdmi audio support
by Yakir Yang 07 Feb '15

07 Feb '15

We found Designware hdmi driver only support audio clock config, we can not play sound through it. To add Designware HDMI Audio support, we make those patch set: 1): fixed dw_hdmi irq bug, add irq control to suspend/resume interfaces. 2): add suspend/resume callback for dw_hdmi rockchip driver. 3): Warp irq control in functions. 4): Combine hdmi_set_clock_regenerator_n() and hdmi_regenerate_cts() 5): Adjust n/cts config order. 6): Set ncts_atomic_write & cts_manual, according to dw_hdmi document. 7): Add audio support for more display resolutions(eg. 800x600). 8): Add audio support for No-CEA display resolutions. 9): Add Audio Sample Channel Status config interfaces to dw_hdmi driver. 10): add audio clock control interfaces o dw_hdmi driver. 11): creat "dw_hdmi-audio" platform device in dw_hdmi driver. 12): add codec driver for hdmi audio, callback dw_hdmi audio config functions. 13): add sound driver for hdmi audio, creat hdmi audio sound card. 14): add dt-bings file and add hdmi_audio node to corresponding dt file. Changes in v3: - Clear Hotplug interrupts before dw_hdmi_fb_register - Wrap irq control in fucntions - Setting the .pm member instead of suspend/resume - Add ID registers parse and record - Combine hdmi_set_clock_regenerator_n() and hdmi_regenerate_cts() - Only adjust the n/cts setting order - Set ncts_atomic_write & cts_manual - Determine whether sample channel should set by desig_id. - Delete hdmi_audio_config interface and modify audio clock control functions - Remove audio_config & get_connect_status callback functions and add write/read/mod register callback functions - Keep audio format config function in dw-hdmi-audio driver and remove audio_config & get_connect_status functions, move jack control to dw-hdmi-audio completely. - Delete the operation of jack in rockchip-hdmi-audio driver, get ready to switch to simple-audio-card driver. - modify cpu-of-node to i2s-controller Changes in v2: - Add irq control to suspend/resume interfaces - Add suspend/resume support for dw_hdmi_rockchip driver - add more n/cts combinations for more display resolutions - enable audio support for No-CEA display mode - Add audio sample channel status setting - Add audio config interfaces to dw_hdmi driver - Update the audio control interfaces - Update dw_hdmi audio control interfaces, and adjust jack report process - give "codec-name" & "codec-dai-name" an const name - remove codec-name and codec-dai-name - rename rockchip,rockchip-hdmi-audio.txt to rockchip,rockchip-dw-hdmi-audio.txt Yakir Yang (15): drm: bridge/dw_hdmi: add irq control to suspend/resume drm: bridge/dw_hdmi: wrap irq control in fucntions drm: rockchip/dw_hdmi_rockchip: add resume/suspend support drm: bridge/dw_hdmi: add indentification registers parse and record drm: bridge/dw_hdmi: combine hdmi_set_clock_regenerator_n() and hdmi_regenerate_cts() drm: bridge/dw_hdmi: adjust n/cts setting order drm: bridge/dw_hdmi: set ncts_atomic_write & cts_manual drm: bridge/dw_hdmi: add audio support for more display resolutions drm: bridge/dw_hdmi: enable audio support for No-CEA display resolutions drm: bridge/dw_hdmi: add audio sample channel status setting drm: bridge/dw_hdmi: add audio clock control interfaces drm: bridge/dw_hdmi: creat dw-hdmi-audio platform device ASoC: dw-hdmi-audio: add codec driver for dw hdmi audio ASoC: rockchip-hdmi-audio: add sound driver for hdmi audio dt-bindings: Add documentation for Rockchip dw-hdmi-audio .../sound/rockchip,rockchip-dw-hdmi-audio.txt | 12 + drivers/gpu/drm/bridge/dw_hdmi.c | 361 ++++++++++++++++++--- drivers/gpu/drm/bridge/dw_hdmi.h | 49 +++ drivers/gpu/drm/rockchip/dw_hdmi-rockchip.c | 16 + include/drm/bridge/dw_hdmi.h | 17 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/dw-hdmi-audio.c | 333 +++++++++++++++++++ sound/soc/codecs/dw-hdmi-audio.h | 79 +++++ sound/soc/rockchip/Kconfig | 9 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rockchip_hdmi_audio.c | 168 ++++++++++ 12 files changed, 1013 insertions(+), 39 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rockchip-dw-hdmi-audio.txt create mode 100644 sound/soc/codecs/dw-hdmi-audio.c create mode 100644 sound/soc/codecs/dw-hdmi-audio.h create mode 100644 sound/soc/rockchip/rockchip_hdmi_audio.c -- 2.1.2

3 19

[alsa-devel] [PATCH 1/4] ASoC: samsung: Remove goni or aquila with the WM8994
by Paul Bolle 07 Feb '15

07 Feb '15

Commit 28c8331d386a ("ARM: S5PV210: Remove support for board files") removed the Kconfig symbols MACH_GONI and MACH_AQUILA. As a result the dependencies of SND_SOC_GONI_AQUILA_WM8994 can never be met. So remove the unbuildable "SoC I2S Audio support for AQUILA/GONI - WM8994". Signed-off-by: Paul Bolle <pebolle(a)tiscali.nl> --- Suggested by Arnd Bergmann (see https://lkml.org/lkml/2014/9/4/381 ). Previously submitted in https://lkml.org/lkml/2014/9/18/128 , but in an awkward way. So this is basically a resend. sound/soc/samsung/Kconfig | 11 -- sound/soc/samsung/Makefile | 2 - sound/soc/samsung/goni_wm8994.c | 289 ---------------------------------------- 3 files changed, 302 deletions(-) delete mode 100644 sound/soc/samsung/goni_wm8994.c diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index fc67f97f19f6..da6d0229cfa2 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -146,17 +146,6 @@ config SND_SOC_SMARTQ select SND_SAMSUNG_I2S select SND_SOC_WM8750 -config SND_SOC_GONI_AQUILA_WM8994 - tristate "SoC I2S Audio support for AQUILA/GONI - WM8994" - depends on SND_SOC_SAMSUNG && (MACH_GONI || MACH_AQUILA) - depends on I2C=y - select SND_SAMSUNG_I2S - select MFD_WM8994 - select SND_SOC_WM8994 - help - Say Y if you want to add support for SoC audio on goni or aquila - with the WM8994. - config SND_SOC_SAMSUNG_SMDK_SPDIF tristate "SoC S/PDIF Audio support for SMDK" depends on SND_SOC_SAMSUNG diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 31e3dba7e3b5..052fe71be518 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -35,7 +35,6 @@ snd-soc-smdk-wm8994-objs := smdk_wm8994.o snd-soc-snow-objs := snow.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o -snd-soc-goni-wm8994-objs := goni_wm8994.o snd-soc-smdk-spdif-objs := smdk_spdif.o snd-soc-smdk-wm8580pcm-objs := smdk_wm8580pcm.o snd-soc-smdk-wm8994pcm-objs := smdk_wm8994pcm.o @@ -63,7 +62,6 @@ obj-$(CONFIG_SND_SOC_SNOW) += snd-soc-snow.o obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_WM9713) += snd-soc-smdk-wm9713.o obj-$(CONFIG_SND_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_SPDIF) += snd-soc-smdk-spdif.o -obj-$(CONFIG_SND_SOC_GONI_AQUILA_WM8994) += snd-soc-goni-wm8994.o obj-$(CONFIG_SND_SOC_SMDK_WM8580_PCM) += snd-soc-smdk-wm8580pcm.o obj-$(CONFIG_SND_SOC_SMDK_WM8994_PCM) += snd-soc-smdk-wm8994pcm.o obj-$(CONFIG_SND_SOC_SPEYSIDE) += snd-soc-speyside.o diff --git a/sound/soc/samsung/goni_wm8994.c b/sound/soc/samsung/goni_wm8994.c deleted file mode 100644 index fad56b9e7369..000000000000 --- a/sound/soc/samsung/goni_wm8994.c +++ /dev/null @@ -1,289 +0,0 @@ -/* - * goni_wm8994.c - * - * Copyright (C) 2010 Samsung Electronics Co.Ltd - * Author: Chanwoo Choi <cw00.choi(a)samsung.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; either version 2 of the License, or (at your - * option) any later version. - * - */ - -#include <linux/module.h> -#include <sound/soc.h> -#include <sound/jack.h> - -#include <asm/mach-types.h> -#include <mach/gpio-samsung.h> - -#include "../codecs/wm8994.h" - -#define MACHINE_NAME 0 -#define CPU_VOICE_DAI 1 - -static const char *aquila_str[] = { - [MACHINE_NAME] = "aquila", - [CPU_VOICE_DAI] = "aquila-voice-dai", -}; - -static struct snd_soc_card goni; -static struct platform_device *goni_snd_device; - -/* 3.5 pie jack */ -static struct snd_soc_jack jack; - -/* 3.5 pie jack detection DAPM pins */ -static struct snd_soc_jack_pin jack_pins[] = { - { - .pin = "Headset Mic", - .mask = SND_JACK_MICROPHONE, - }, { - .pin = "Headset Stereophone", - .mask = SND_JACK_HEADPHONE | SND_JACK_MECHANICAL | - SND_JACK_AVOUT, - }, -}; - -/* 3.5 pie jack detection gpios */ -static struct snd_soc_jack_gpio jack_gpios[] = { - { - .gpio = S5PV210_GPH0(6), - .name = "DET_3.5", - .report = SND_JACK_HEADSET | SND_JACK_MECHANICAL | - SND_JACK_AVOUT, - .debounce_time = 200, - }, -}; - -static const struct snd_soc_dapm_widget goni_dapm_widgets[] = { - SND_SOC_DAPM_SPK("Ext Left Spk", NULL), - SND_SOC_DAPM_SPK("Ext Right Spk", NULL), - SND_SOC_DAPM_SPK("Ext Rcv", NULL), - SND_SOC_DAPM_HP("Headset Stereophone", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_MIC("Main Mic", NULL), - SND_SOC_DAPM_MIC("2nd Mic", NULL), - SND_SOC_DAPM_LINE("Radio In", NULL), -}; - -static const struct snd_soc_dapm_route goni_dapm_routes[] = { - {"Ext Left Spk", NULL, "SPKOUTLP"}, - {"Ext Left Spk", NULL, "SPKOUTLN"}, - - {"Ext Right Spk", NULL, "SPKOUTRP"}, - {"Ext Right Spk", NULL, "SPKOUTRN"}, - - {"Ext Rcv", NULL, "HPOUT2N"}, - {"Ext Rcv", NULL, "HPOUT2P"}, - - {"Headset Stereophone", NULL, "HPOUT1L"}, - {"Headset Stereophone", NULL, "HPOUT1R"}, - - {"IN1RN", NULL, "Headset Mic"}, - {"IN1RP", NULL, "Headset Mic"}, - - {"IN1RN", NULL, "2nd Mic"}, - {"IN1RP", NULL, "2nd Mic"}, - - {"IN1LN", NULL, "Main Mic"}, - {"IN1LP", NULL, "Main Mic"}, - - {"IN2LN", NULL, "Radio In"}, - {"IN2RN", NULL, "Radio In"}, -}; - -static int goni_wm8994_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret; - - /* set endpoints to not connected */ - snd_soc_dapm_nc_pin(dapm, "IN2LP:VXRN"); - snd_soc_dapm_nc_pin(dapm, "IN2RP:VXRP"); - snd_soc_dapm_nc_pin(dapm, "LINEOUT1N"); - snd_soc_dapm_nc_pin(dapm, "LINEOUT1P"); - snd_soc_dapm_nc_pin(dapm, "LINEOUT2N"); - snd_soc_dapm_nc_pin(dapm, "LINEOUT2P"); - - if (machine_is_aquila()) { - snd_soc_dapm_nc_pin(dapm, "SPKOUTRN"); - snd_soc_dapm_nc_pin(dapm, "SPKOUTRP"); - } - - /* Headset jack detection */ - ret = snd_soc_jack_new(codec, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_MECHANICAL | SND_JACK_AVOUT, - &jack); - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(&jack, ARRAY_SIZE(jack_pins), jack_pins); - if (ret) - return ret; - - ret = snd_soc_jack_add_gpios(&jack, ARRAY_SIZE(jack_gpios), jack_gpios); - if (ret) - return ret; - - return 0; -} - -static int goni_hifi_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; - unsigned int pll_out = 24000000; - int ret = 0; - - /* set the codec FLL */ - ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, 0, pll_out, - params_rate(params) * 256); - if (ret < 0) - return ret; - - /* set the codec system clock */ - ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL1, - params_rate(params) * 256, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - return 0; -} - -static struct snd_soc_ops goni_hifi_ops = { - .hw_params = goni_hifi_hw_params, -}; - -static int goni_voice_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; - unsigned int pll_out = 24000000; - int ret = 0; - - if (params_rate(params) != 8000) - return -EINVAL; - - /* set the codec FLL */ - ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL2, 0, pll_out, - params_rate(params) * 256); - if (ret < 0) - return ret; - - /* set the codec system clock */ - ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL2, - params_rate(params) * 256, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - return 0; -} - -static struct snd_soc_dai_driver voice_dai = { - .name = "goni-voice-dai", - .id = 0, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, -}; - -static const struct snd_soc_component_driver voice_component = { - .name = "goni-voice", -}; - -static struct snd_soc_ops goni_voice_ops = { - .hw_params = goni_voice_hw_params, -}; - -static struct snd_soc_dai_link goni_dai[] = { -{ - .name = "WM8994", - .stream_name = "WM8994 HiFi", - .cpu_dai_name = "samsung-i2s.0", - .codec_dai_name = "wm8994-aif1", - .platform_name = "samsung-i2s.0", - .codec_name = "wm8994-codec.0-001a", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, - .init = goni_wm8994_init, - .ops = &goni_hifi_ops, -}, { - .name = "WM8994 Voice", - .stream_name = "Voice", - .cpu_dai_name = "goni-voice-dai", - .codec_dai_name = "wm8994-aif2", - .codec_name = "wm8994-codec.0-001a", - .dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_IB_IF | - SND_SOC_DAIFMT_CBM_CFM, - .ops = &goni_voice_ops, -}, -}; - -static struct snd_soc_card goni = { - .name = "goni", - .owner = THIS_MODULE, - .dai_link = goni_dai, - .num_links = ARRAY_SIZE(goni_dai), - - .dapm_widgets = goni_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(goni_dapm_widgets), - .dapm_routes = goni_dapm_routes, - .num_dapm_routes = ARRAY_SIZE(goni_dapm_routes), -}; - -static int __init goni_init(void) -{ - int ret; - - if (machine_is_aquila()) { - voice_dai.name = aquila_str[CPU_VOICE_DAI]; - goni_dai[1].cpu_dai_name = aquila_str[CPU_VOICE_DAI]; - goni.name = aquila_str[MACHINE_NAME]; - } else if (!machine_is_goni()) - return -ENODEV; - - goni_snd_device = platform_device_alloc("soc-audio", -1); - if (!goni_snd_device) - return -ENOMEM; - - /* register voice DAI here */ - ret = devm_snd_soc_register_component(&goni_snd_device->dev, - &voice_component, &voice_dai, 1); - if (ret) { - platform_device_put(goni_snd_device); - return ret; - } - - platform_set_drvdata(goni_snd_device, &goni); - ret = platform_device_add(goni_snd_device); - - if (ret) - platform_device_put(goni_snd_device); - - return ret; -} - -static void __exit goni_exit(void) -{ - platform_device_unregister(goni_snd_device); -} - -module_init(goni_init); -module_exit(goni_exit); - -/* Module information */ -MODULE_DESCRIPTION("ALSA SoC WM8994 GONI(S5PV210)"); -MODULE_AUTHOR("Chanwoo Choi <cw00.choi(a)samsung.com>"); -MODULE_LICENSE("GPL"); -- 1.9.3

2 1

[alsa-devel] [PATCH 4/4] ASoC: samsung: remove outdated Kconfig references
by Paul Bolle 07 Feb '15

07 Feb '15

The Kconfig symbols MACH_SMDKC110 and MACH_SMDKV210 were removed in v3.17. But the Kconfig entries for SND_SOC_SAMSUNG_SMDK_WM8580 and SND_SOC_SAMSUNG_SMDK_WM9713 still reference them. Perhaps these references to those removed symbols should be changed into references to an existing symbol (such as MACH_S5PV210_DT). But apparently no one noticed that these references were pointless ever since v3.17. So it seems they can simply be removed. Let's do that now. Signed-off-by: Paul Bolle <pebolle(a)tiscali.nl> --- sound/soc/samsung/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 5f983d064920..01980bb2c6e7 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -53,7 +53,7 @@ config SND_SOC_SAMSUNG_JIVE_WM8750 config SND_SOC_SAMSUNG_SMDK_WM8580 tristate "SoC I2S Audio support for WM8580 on SMDK" - depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKV210 || MACH_SMDKC110) + depends on SND_SOC_SAMSUNG && MACH_SMDK6410 depends on REGMAP_I2C select SND_SOC_WM8580 select SND_SAMSUNG_I2S @@ -134,7 +134,7 @@ config SND_SOC_SAMSUNG_RX1950_UDA1380 config SND_SOC_SAMSUNG_SMDK_WM9713 tristate "SoC AC97 Audio support for SMDK with WM9713" - depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKV210 || MACH_SMDKC110) + depends on SND_SOC_SAMSUNG && MACH_SMDK6410 select SND_SOC_WM9713 select SND_SAMSUNG_AC97 help -- 1.9.3

2 1

[alsa-devel] [PATCH 3/4] ASoC: samsung: MACH_SMDKC100 isn't supported anymore
by Paul Bolle 07 Feb '15

07 Feb '15

Commit b8529ec1c1b0 ("ARM: S5PC100: no more support S5PC100 SoC") removed Kconfig symbol MACH_SMDKC100. But there are still two references to that symbol left. These are now pointless. Remove them too. Signed-off-by: Paul Bolle <pebolle(a)tiscali.nl> --- Apparently Kukjin had a fixup queued (see http://mailman.alsa-project.org/pipermail/alsa-devel/2014-September/081482.… ). But that fixup never entered the tree, so let's try again. sound/soc/samsung/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 3be649098b30..5f983d064920 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -53,7 +53,7 @@ config SND_SOC_SAMSUNG_JIVE_WM8750 config SND_SOC_SAMSUNG_SMDK_WM8580 tristate "SoC I2S Audio support for WM8580 on SMDK" - depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110) + depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKV210 || MACH_SMDKC110) depends on REGMAP_I2C select SND_SOC_WM8580 select SND_SAMSUNG_I2S @@ -134,7 +134,7 @@ config SND_SOC_SAMSUNG_RX1950_UDA1380 config SND_SOC_SAMSUNG_SMDK_WM9713 tristate "SoC AC97 Audio support for SMDK with WM9713" - depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110) + depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKV210 || MACH_SMDKC110) select SND_SOC_WM9713 select SND_SAMSUNG_AC97 help -- 1.9.3

2 1

[alsa-devel] [PATCH 2/4] ASoC: samsung: Remove PCM support for WM8580 on SMDK
by Paul Bolle 07 Feb '15

07 Feb '15

Commit 28c8331d386a ("ARM: S5PV210: Remove support for board files") removed the Kconfig symbols MACH_SMDKC110 and MACH_SMDKV210. As a result the dependencies of SND_SOC_SMDK_WM8580_PCM can never be met. So remove the unbuildable "SoC PCM Audio support for WM8580 on SMDK". Signed-off-by: Paul Bolle <pebolle(a)tiscali.nl> --- Suggested by Arnd Bergmann (see https://lkml.org/lkml/2014/9/4/381 ). Previously submitted in https://lkml.org/lkml/2014/9/18/147 , but again in an awkward way. So this is a resend too. sound/soc/samsung/Kconfig | 9 -- sound/soc/samsung/Makefile | 2 - sound/soc/samsung/smdk_wm8580pcm.c | 176 ------------------------------------- 3 files changed, 187 deletions(-) delete mode 100644 sound/soc/samsung/smdk_wm8580pcm.c diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index da6d0229cfa2..3be649098b30 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -153,15 +153,6 @@ config SND_SOC_SAMSUNG_SMDK_SPDIF help Say Y if you want to add support for SoC S/PDIF audio on the SMDK. -config SND_SOC_SMDK_WM8580_PCM - tristate "SoC PCM Audio support for WM8580 on SMDK" - depends on SND_SOC_SAMSUNG && (MACH_SMDKV210 || MACH_SMDKC110) - depends on REGMAP_I2C - select SND_SOC_WM8580 - select SND_SAMSUNG_PCM - help - Say Y if you want to add support for SoC audio on the SMDK. - config SND_SOC_SMDK_WM8994_PCM tristate "SoC PCM Audio support for WM8994 on SMDK" depends on SND_SOC_SAMSUNG diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 052fe71be518..2b32c371da55 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -36,7 +36,6 @@ snd-soc-snow-objs := snow.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o snd-soc-smdk-spdif-objs := smdk_spdif.o -snd-soc-smdk-wm8580pcm-objs := smdk_wm8580pcm.o snd-soc-smdk-wm8994pcm-objs := smdk_wm8994pcm.o snd-soc-speyside-objs := speyside.o snd-soc-tobermory-objs := tobermory.o @@ -62,7 +61,6 @@ obj-$(CONFIG_SND_SOC_SNOW) += snd-soc-snow.o obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_WM9713) += snd-soc-smdk-wm9713.o obj-$(CONFIG_SND_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_SPDIF) += snd-soc-smdk-spdif.o -obj-$(CONFIG_SND_SOC_SMDK_WM8580_PCM) += snd-soc-smdk-wm8580pcm.o obj-$(CONFIG_SND_SOC_SMDK_WM8994_PCM) += snd-soc-smdk-wm8994pcm.o obj-$(CONFIG_SND_SOC_SPEYSIDE) += snd-soc-speyside.o obj-$(CONFIG_SND_SOC_TOBERMORY) += snd-soc-tobermory.o diff --git a/sound/soc/samsung/smdk_wm8580pcm.c b/sound/soc/samsung/smdk_wm8580pcm.c deleted file mode 100644 index 6deec5234c92..000000000000 --- a/sound/soc/samsung/smdk_wm8580pcm.c +++ /dev/null @@ -1,176 +0,0 @@ -/* - * sound/soc/samsung/smdk_wm8580pcm.c - * - * Copyright (c) 2011 Samsung Electronics Co. Ltd - * - * 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; either version 2 of the License, or (at your - * option) any later version. - */ -#include <linux/module.h> -#include <sound/soc.h> -#include <sound/pcm_params.h> -#include <sound/pcm.h> - -#include <asm/mach-types.h> - -#include "../codecs/wm8580.h" -#include "dma.h" -#include "pcm.h" - -/* - * Board Settings: - * o '1' means 'ON' - * o '0' means 'OFF' - * o 'X' means 'Don't care' - * - * SMDK6410 Base B/D: CFG1-0000, CFG2-1111 - * SMDKC110, SMDKV210: CFGB11-100100, CFGB12-0000 - */ - -#define SMDK_WM8580_EXT_OSC 12000000 -#define SMDK_WM8580_EXT_MCLK 4096000 -#define SMDK_WM8580_EXT_VOICE 2048000 - -static unsigned long mclk_freq; -static unsigned long xtal_freq; - -/* - * If MCLK clock directly gets from XTAL, we don't have to use PLL - * to make MCLK, but if XTAL clock source connects with other codec - * pin (like XTI), we should have to set codec's PLL to make MCLK. - * Because Samsung SoC does not support pcmcdclk output like I2S. - */ - -static int smdk_wm8580_pcm_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; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int rfs, ret; - - switch (params_rate(params)) { - case 8000: - break; - default: - printk(KERN_ERR "%s:%d Sampling Rate %u not supported!\n", - __func__, __LINE__, params_rate(params)); - return -EINVAL; - } - - rfs = mclk_freq / params_rate(params) / 2; - - if (mclk_freq == xtal_freq) { - ret = snd_soc_dai_set_sysclk(codec_dai, WM8580_CLKSRC_MCLK, - mclk_freq, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_MCLK, - WM8580_CLKSRC_MCLK); - if (ret < 0) - return ret; - } else { - ret = snd_soc_dai_set_sysclk(codec_dai, WM8580_CLKSRC_PLLA, - mclk_freq, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_MCLK, - WM8580_CLKSRC_PLLA); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_pll(codec_dai, WM8580_PLLA, 0, - xtal_freq, mclk_freq); - if (ret < 0) - return ret; - } - - /* Set PCM source clock on CPU */ - ret = snd_soc_dai_set_sysclk(cpu_dai, S3C_PCM_CLKSRC_MUX, - mclk_freq, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* Set SCLK_DIV for making bclk */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_PCM_SCLK_PER_FS, rfs); - if (ret < 0) - return ret; - - return 0; -} - -static struct snd_soc_ops smdk_wm8580_pcm_ops = { - .hw_params = smdk_wm8580_pcm_hw_params, -}; - -#define SMDK_DAI_FMT (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF | \ - SND_SOC_DAIFMT_CBS_CFS) - -static struct snd_soc_dai_link smdk_dai[] = { - { - .name = "WM8580 PAIF PCM RX", - .stream_name = "Playback", - .cpu_dai_name = "samsung-pcm.0", - .codec_dai_name = "wm8580-hifi-playback", - .platform_name = "samsung-audio", - .codec_name = "wm8580.0-001b", - .dai_fmt = SMDK_DAI_FMT, - .ops = &smdk_wm8580_pcm_ops, - }, { - .name = "WM8580 PAIF PCM TX", - .stream_name = "Capture", - .cpu_dai_name = "samsung-pcm.0", - .codec_dai_name = "wm8580-hifi-capture", - .platform_name = "samsung-pcm.0", - .codec_name = "wm8580.0-001b", - .dai_fmt = SMDK_DAI_FMT, - .ops = &smdk_wm8580_pcm_ops, - }, -}; - -static struct snd_soc_card smdk_pcm = { - .name = "SMDK-PCM", - .owner = THIS_MODULE, - .dai_link = smdk_dai, - .num_links = 2, -}; - -/* - * After SMDKC110 Base Board's Rev is '0.1', 12MHz External OSC(X1) - * is absent (or not connected), so we connect EXT_VOICE_CLK(OSC4), - * 2.0484Mhz, directly with MCLK both Codec and SoC. - */ -static int snd_smdk_probe(struct platform_device *pdev) -{ - int ret = 0; - - xtal_freq = SMDK_WM8580_EXT_OSC; - mclk_freq = SMDK_WM8580_EXT_MCLK; - - if (machine_is_smdkc110() || machine_is_smdkv210()) - xtal_freq = mclk_freq = SMDK_WM8580_EXT_VOICE; - - smdk_pcm.dev = &pdev->dev; - ret = devm_snd_soc_register_card(&pdev->dev, &smdk_pcm); - if (ret) - dev_err(&pdev->dev, "snd_soc_register_card failed %d\n", ret); - - return ret; -} - -static struct platform_driver snd_smdk_driver = { - .driver = { - .name = "samsung-smdk-pcm", - }, - .probe = snd_smdk_probe, -}; - -module_platform_driver(snd_smdk_driver); - -MODULE_AUTHOR("Sangbeom Kim, <sbkim73(a)samsung.com>"); -MODULE_DESCRIPTION("ALSA SoC SMDK WM8580 for PCM"); -MODULE_LICENSE("GPL"); -- 1.9.3

2 1

[alsa-devel] [PATCH v2] soc: intel: sst-haswell-ipc: Remove unused functions
by Rickard Strandqvist 07 Feb '15

07 Feb '15

Removes some functions that are not used anywhere: sst_hsw_stream_unmute() sst_hsw_stream_mute() msg_set_stage_type() sst_hsw_dx_get_state() sst_hsw_stream_set_write_position() sst_hsw_stream_get_vol_reg() sst_hsw_stream_get_peak_reg() sst_hsw_stream_get_pointer_reg() sst_hsw_stream_get_read_reg() sst_hsw_stream_get_mixer_id() sst_hsw_stream_get_hw_id() sst_hsw_mixer_set_volume_curve() sst_hsw_mixer_unmute() sst_hsw_mixer_mute() sst_hsw_stream_set_volume_curve() This was partially found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/soc/intel/sst-haswell-ipc.c | 168 ------------------------------------- sound/soc/intel/sst-haswell-ipc.h | 31 ------- 2 files changed, 199 deletions(-) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 3f8c482..d693d4a 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -337,12 +337,6 @@ static inline u32 msg_get_stage_type(u32 msg) return (msg & IPC_STG_TYPE_MASK) >> IPC_STG_TYPE_SHIFT; } -static inline u32 msg_set_stage_type(u32 msg, u32 type) -{ - return (msg & ~IPC_STG_TYPE_MASK) + - (type << IPC_STG_TYPE_SHIFT); -} - static inline u32 msg_get_stream_id(u32 msg) { return (msg & IPC_STR_ID_MASK) >> IPC_STR_ID_SHIFT; @@ -969,45 +963,6 @@ int sst_hsw_fw_get_version(struct sst_hsw *hsw, } /* Mixer Controls */ -int sst_hsw_stream_mute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, - u32 stage_id, u32 channel) -{ - int ret; - - ret = sst_hsw_stream_get_volume(hsw, stream, stage_id, channel, - &stream->mute_volume[channel]); - if (ret < 0) - return ret; - - ret = sst_hsw_stream_set_volume(hsw, stream, stage_id, channel, 0); - if (ret < 0) { - dev_err(hsw->dev, "error: can't unmute stream %d channel %d\n", - stream->reply.stream_hw_id, channel); - return ret; - } - - stream->mute[channel] = 1; - return 0; -} - -int sst_hsw_stream_unmute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, - u32 stage_id, u32 channel) - -{ - int ret; - - stream->mute[channel] = 0; - ret = sst_hsw_stream_set_volume(hsw, stream, stage_id, channel, - stream->mute_volume[channel]); - if (ret < 0) { - dev_err(hsw->dev, "error: can't unmute stream %d channel %d\n", - stream->reply.stream_hw_id, channel); - return ret; - } - - return 0; -} - int sst_hsw_stream_get_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 *volume) { @@ -1021,17 +976,6 @@ int sst_hsw_stream_get_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream return 0; } -int sst_hsw_stream_set_volume_curve(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u64 curve_duration, - enum sst_hsw_volume_curve curve) -{ - /* curve duration in steps of 100ns */ - stream->vol_req.curve_duration = curve_duration; - stream->vol_req.curve_type = curve; - - return 0; -} - /* stream volume */ int sst_hsw_stream_set_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 volume) @@ -1083,42 +1027,6 @@ int sst_hsw_stream_set_volume(struct sst_hsw *hsw, return 0; } -int sst_hsw_mixer_mute(struct sst_hsw *hsw, u32 stage_id, u32 channel) -{ - int ret; - - ret = sst_hsw_mixer_get_volume(hsw, stage_id, channel, - &hsw->mute_volume[channel]); - if (ret < 0) - return ret; - - ret = sst_hsw_mixer_set_volume(hsw, stage_id, channel, 0); - if (ret < 0) { - dev_err(hsw->dev, "error: failed to unmute mixer channel %d\n", - channel); - return ret; - } - - hsw->mute[channel] = 1; - return 0; -} - -int sst_hsw_mixer_unmute(struct sst_hsw *hsw, u32 stage_id, u32 channel) -{ - int ret; - - ret = sst_hsw_mixer_set_volume(hsw, stage_id, channel, - hsw->mixer_info.volume_register_address[channel]); - if (ret < 0) { - dev_err(hsw->dev, "error: failed to unmute mixer channel %d\n", - channel); - return ret; - } - - hsw->mute[channel] = 0; - return 0; -} - int sst_hsw_mixer_get_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, u32 *volume) { @@ -1132,16 +1040,6 @@ int sst_hsw_mixer_get_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, return 0; } -int sst_hsw_mixer_set_volume_curve(struct sst_hsw *hsw, - u64 curve_duration, enum sst_hsw_volume_curve curve) -{ - /* curve duration in steps of 100ns */ - hsw->curve_duration = curve_duration; - hsw->curve_type = curve; - - return 0; -} - /* global mixer volume */ int sst_hsw_mixer_set_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, u32 volume) @@ -1434,48 +1332,6 @@ int sst_hsw_stream_commit(struct sst_hsw *hsw, struct sst_hsw_stream *stream) /* Stream Information - these calls could be inline but we want the IPC ABI to be opaque to client PCM drivers to cope with any future ABI changes */ -int sst_hsw_stream_get_hw_id(struct sst_hsw *hsw, - struct sst_hsw_stream *stream) -{ - return stream->reply.stream_hw_id; -} - -int sst_hsw_stream_get_mixer_id(struct sst_hsw *hsw, - struct sst_hsw_stream *stream) -{ - return stream->reply.mixer_hw_id; -} - -u32 sst_hsw_stream_get_read_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream) -{ - return stream->reply.read_position_register_address; -} - -u32 sst_hsw_stream_get_pointer_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream) -{ - return stream->reply.presentation_position_register_address; -} - -u32 sst_hsw_stream_get_peak_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 channel) -{ - if (channel >= 2) - return 0; - - return stream->reply.peak_meter_register_address[channel]; -} - -u32 sst_hsw_stream_get_vol_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 channel) -{ - if (channel >= 2) - return 0; - - return stream->reply.volume_register_address[channel]; -} - int sst_hsw_mixer_get_info(struct sst_hsw *hsw) { struct sst_hsw_ipc_stream_info_reply *reply; @@ -1598,30 +1454,6 @@ u64 sst_hsw_get_dsp_presentation_position(struct sst_hsw *hsw, return ppos; } -int sst_hsw_stream_set_write_position(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 stage_id, u32 position) -{ - u32 header; - int ret; - - trace_stream_write_position(stream->reply.stream_hw_id, position); - - header = IPC_GLB_TYPE(IPC_GLB_STREAM_MESSAGE) | - IPC_STR_TYPE(IPC_STR_STAGE_MESSAGE); - header |= (stream->reply.stream_hw_id << IPC_STR_ID_SHIFT); - header |= (IPC_STG_SET_WRITE_POSITION << IPC_STG_TYPE_SHIFT); - header |= (stage_id << IPC_STG_ID_SHIFT); - stream->wpos.position = position; - - ret = ipc_tx_message_nowait(hsw, header, &stream->wpos, - sizeof(stream->wpos)); - if (ret < 0) - dev_err(hsw->dev, "error: stream %d set position %d failed\n", - stream->reply.stream_hw_id, position); - - return ret; -} - /* physical BE config */ int sst_hsw_device_set_config(struct sst_hsw *hsw, enum sst_hsw_device_id dev, enum sst_hsw_device_mclk mclk, diff --git a/sound/soc/intel/sst-haswell-ipc.h b/sound/soc/intel/sst-haswell-ipc.h index 138e894..c1ad901 100644 --- a/sound/soc/intel/sst-haswell-ipc.h +++ b/sound/soc/intel/sst-haswell-ipc.h @@ -376,32 +376,17 @@ int sst_hsw_fw_get_version(struct sst_hsw *hsw, u32 create_channel_map(enum sst_hsw_channel_config config); /* Stream Mixer Controls - */ -int sst_hsw_stream_mute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, - u32 stage_id, u32 channel); -int sst_hsw_stream_unmute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, - u32 stage_id, u32 channel); - int sst_hsw_stream_set_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 volume); int sst_hsw_stream_get_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 *volume); -int sst_hsw_stream_set_volume_curve(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u64 curve_duration, - enum sst_hsw_volume_curve curve); - /* Global Mixer Controls - */ -int sst_hsw_mixer_mute(struct sst_hsw *hsw, u32 stage_id, u32 channel); -int sst_hsw_mixer_unmute(struct sst_hsw *hsw, u32 stage_id, u32 channel); - int sst_hsw_mixer_set_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, u32 volume); int sst_hsw_mixer_get_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, u32 *volume); -int sst_hsw_mixer_set_volume_curve(struct sst_hsw *hsw, - u64 curve_duration, enum sst_hsw_volume_curve curve); - /* Stream API */ struct sst_hsw_stream *sst_hsw_stream_new(struct sst_hsw *hsw, int id, u32 (*get_write_position)(struct sst_hsw_stream *stream, void *data), @@ -440,18 +425,6 @@ int sst_hsw_stream_set_pmemory_info(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 offset, u32 size); int sst_hsw_stream_set_smemory_info(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 offset, u32 size); -int sst_hsw_stream_get_hw_id(struct sst_hsw *hsw, - struct sst_hsw_stream *stream); -int sst_hsw_stream_get_mixer_id(struct sst_hsw *hsw, - struct sst_hsw_stream *stream); -u32 sst_hsw_stream_get_read_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream); -u32 sst_hsw_stream_get_pointer_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream); -u32 sst_hsw_stream_get_peak_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 channel); -u32 sst_hsw_stream_get_vol_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 channel); int sst_hsw_mixer_get_info(struct sst_hsw *hsw); /* Stream ALSA trigger operations */ @@ -466,8 +439,6 @@ int sst_hsw_stream_get_read_pos(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 *position); int sst_hsw_stream_get_write_pos(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 *position); -int sst_hsw_stream_set_write_position(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 stage_id, u32 position); u32 sst_hsw_get_dsp_position(struct sst_hsw *hsw, struct sst_hsw_stream *stream); u64 sst_hsw_get_dsp_presentation_position(struct sst_hsw *hsw, @@ -481,8 +452,6 @@ int sst_hsw_device_set_config(struct sst_hsw *hsw, /* DX Config */ int sst_hsw_dx_set_state(struct sst_hsw *hsw, enum sst_hsw_dx_state state, struct sst_hsw_ipc_dx_reply *dx); -int sst_hsw_dx_get_state(struct sst_hsw *hsw, u32 item, - u32 *offset, u32 *size, u32 *source); /* init */ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata); -- 1.7.10.4

2 1

[alsa-devel] [PATCH v3 alsa-lib 1/3] snd_pcm_start_at and friends.
by Tim Cussins 06 Feb '15

06 Feb '15

Upcoming audio hardware has the ability to begin playback when the value of a high-precision free-running counter matches some programmable value. This patch exposes such capability in a general way, allowing for several clock sources, including both system clocks and audio hardware clocks. We define 3 new methods that operate on a snd_pcm_t, and 1 that operates on a snd_pcm_status_t. - snd_pcm_start_at() allows client code to delegate the starting of the stream to the kernel. - snd_pcm_start_at_abort() allows client code to cancel a previous call to snd_pcm_start_at(). - snd_pcm_start_at_gettime() allows client code to query the current time for a given clock. - snd_pcm_status_get_startat_state() allows client code to extract information about any pending start_at timer from a snd_pcm_status_t object. We define a new enum: snd_pcm_startat_clock_type_t, which reinforces a clean boundary between the capabilities of the timestamping system and the start_at system. The clock and audio hw sources are grouped together because the difference is irrelevant to the use of snd_pcm_start_at(). Signed-off-by: Tim Cussins <timcussins(a)eml.cc> diff --git a/include/pcm.h b/include/pcm.h index 0655e7f..4f4834b 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -330,6 +330,17 @@ typedef enum _snd_pcm_tstamp_type { SND_PCM_TSTAMP_TYPE_LAST = SND_PCM_TSTAMP_TYPE_MONOTONIC_RAW, } snd_pcm_tstamp_type_t; +/** PCM start_at clock types */ +typedef enum _snd_pcm_startat_clock_type { + /** POSIX CLOCK_REALTIME equivalent */ + SND_PCM_STARTAT_CLOCK_TYPE_GETTIMEOFDAY = 0, + /** POSIX CLOCK_MONOTONIC equivalent */ + SND_PCM_STARTAT_CLOCK_TYPE_MONOTONIC, + /** Default link clock */ + SND_PCM_STARTAT_CLOCK_TYPE_LINK, + SND_PCM_STARTAT_CLOCK_TYPE_LAST = SND_PCM_STARTAT_CLOCK_TYPE_LINK, +} snd_pcm_startat_clock_type_t; + /** Unsigned frames quantity */ typedef unsigned long snd_pcm_uframes_t; /** Signed frames quantity */ @@ -478,6 +489,9 @@ int snd_pcm_prepare(snd_pcm_t *pcm); int snd_pcm_reset(snd_pcm_t *pcm); int snd_pcm_status(snd_pcm_t *pcm, snd_pcm_status_t *status); int snd_pcm_start(snd_pcm_t *pcm); +int snd_pcm_start_at(snd_pcm_t *pcm, snd_pcm_startat_clock_type_t clock_type, const snd_htimestamp_t* start_time); +int snd_pcm_start_at_abort(snd_pcm_t *pcm); +int snd_pcm_start_at_gettime(snd_pcm_t *pcm, snd_pcm_startat_clock_type_t clock_type, snd_htimestamp_t* current_time); int snd_pcm_drop(snd_pcm_t *pcm); int snd_pcm_drain(snd_pcm_t *pcm); int snd_pcm_pause(snd_pcm_t *pcm, int enable); @@ -984,6 +998,7 @@ snd_pcm_sframes_t snd_pcm_status_get_delay(const snd_pcm_status_t *obj); snd_pcm_uframes_t snd_pcm_status_get_avail(const snd_pcm_status_t *obj); snd_pcm_uframes_t snd_pcm_status_get_avail_max(const snd_pcm_status_t *obj); snd_pcm_uframes_t snd_pcm_status_get_overrange(const snd_pcm_status_t *obj); +void snd_pcm_status_get_startat_state(const snd_pcm_status_t *obj, int *pending, snd_pcm_startat_clock_type_t *clock_type, snd_htimestamp_t *start_time); /** \} */ diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index e74e02f..f2cbd8e 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -1085,6 +1085,74 @@ int snd_pcm_start(snd_pcm_t *pcm) } /** + * \brief Start a PCM at a specified point in the future + * \param pcm PCM handle + * \param clock_type Specifies the start_at clock with which to interpret \p start_time + * \param start_time Absolute time at which to start the stream + * \return 0 on success otherwise a negative error code + * \retval -ENOSYS operation not supported for the current timestamp type + * \retval -EINVAL timespec, tstamp_class or tstamp_type is invalid + * \retval -ETIME requested start_time cannot be satisfied + * + * This method is non-blocking: It establishes an appropriate timer in the kernel + * that will start the stream on expiry. + * + * Any pending timer is unconditionally cancelled. + */ +int snd_pcm_start_at(snd_pcm_t *pcm, snd_pcm_startat_clock_type_t clock_type, const snd_htimestamp_t *start_time) +{ + assert(pcm); + assert(start_time); + if (CHECK_SANITY(! pcm->setup)) { + SNDMSG("PCM not set up"); + return -EIO; + } + if (pcm->fast_ops->start_at) { + return pcm->fast_ops->start_at(pcm->fast_op_arg, clock_type, start_time); + } + return -EINVAL; +} + +/** + * \brief Abort the pending PCM start_at timer + * \param pcm PCM handle + * \return 0 on success otherwise a negative error code + * + * If a start_at timer is pending, it is cancelled. + */ +int snd_pcm_start_at_abort(snd_pcm_t *pcm) +{ + assert(pcm); + if (CHECK_SANITY(! pcm->setup)) { + SNDMSG("PCM not set up"); + return -EIO; + } + if (pcm->fast_ops->start_at_abort) { + return pcm->fast_ops->start_at_abort(pcm->fast_op_arg); + } + return -EINVAL; +} + +/** + * \brief Get current time for a given start_at clock + * \param pcm PCM handle + * \param clock_type Start_at clock type e.g. SND_PCM_STARTAT_CLOCK_TYPE_GETTIMEOFDAY + * \return 0 on success otherwise a negative error code + */ +int snd_pcm_start_at_gettime(snd_pcm_t *pcm, snd_pcm_startat_clock_type_t clock_type, snd_htimestamp_t *current_time) +{ + assert(pcm); + if (CHECK_SANITY(! pcm->setup)) { + SNDMSG("PCM not set up"); + return -EIO; + } + if (pcm->fast_ops->start_at_gettime) { + return pcm->fast_ops->start_at_gettime(pcm->fast_op_arg, clock_type, current_time); + } + return -EINVAL; +} + +/** * \brief Stop a PCM dropping pending frames * \param pcm PCM handle * \return 0 on success otherwise a negative error code @@ -6380,6 +6448,20 @@ snd_pcm_uframes_t snd_pcm_status_get_overrange(const snd_pcm_status_t *obj) } /** + * \brief Get status of a pending start_at timer, if any. + * \param pending output param: 1 if start_at timer pending, 0 otherwise + * \param clock_type output param: if pending == 1, is updated with current start_at timer clock_type + * \param start_time output param: if pending == 1, is updated with current start_at timer start_time + */ +void snd_pcm_status_get_startat_state(const snd_pcm_status_t *obj, int *pending, snd_pcm_startat_clock_type_t *clock_type, snd_htimestamp_t *start_time) +{ + assert(obj); + *pending = obj->start_at_pending; + *clock_type = obj->start_at_clock_type; + *start_time = obj->start_at_start_time; +} + +/** * \brief get size of #snd_pcm_info_t * \return size in bytes */ -- 1.9.1

1 2

[alsa-devel] [PATCH 2/2] davinci-mcasp: add option to use AHCLKX pin as clock in
by Urs Fässler 06 Feb '15

06 Feb '15

Signed-off-by: Urs Fässler <urs.fassler(a)bytesatwork.ch> --- .../bindings/sound/davinci-mcasp-audio.txt | 1 + sound/soc/davinci/davinci-mcasp.c | 26 +++++++++++++++++++--- 2 files changed, 24 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt index 46bc982..22a91d2 100644 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt @@ -39,6 +39,7 @@ Optional properties: please refer to pinctrl-bindings.txt - fck_parent : Should contain a valid clock name which will be used as parent for the McASP fck +- ahclkx-pin : if set use AHCLKX pin for clock in instead of ACLKX Example: diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 396cdec..3730288 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -85,6 +85,7 @@ struct davinci_mcasp { int sysclk_freq; bool bclk_master; + bool ahclkx_pin; /* McASP FIFO related */ u8 txnumevt; @@ -429,7 +430,12 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, break; case SND_SOC_DAIFMT_CBM_CFS: /* codec is clock master and frame slave */ - mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + if (mcasp->ahclkx_pin) + mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, + ACLKXE); + else + mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, + ACLKXE); mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, AFSXE); mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); @@ -441,7 +447,12 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, break; case SND_SOC_DAIFMT_CBM_CFM: /* codec is clock and frame master */ - mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + if (mcasp->ahclkx_pin) + mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, + ACLKXE); + else + mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, + ACLKXE); mcasp_clr_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, AFSXE); mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); @@ -542,7 +553,12 @@ static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id, struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); if (mcasp->bclk_master) { - mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE); + if (mcasp->ahclkx_pin) + mcasp_clr_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, + AHCLKXE); + else + mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, + AHCLKXE); mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE); mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AHCLKX); } else { @@ -1340,6 +1356,8 @@ static struct davinci_mcasp_pdata *davinci_mcasp_set_pdata_from_of( if (ret >= 0) pdata->sram_size_capture = val; + pdata->clk_input_pin = of_property_read_bool(np, "ahclkx-pin"); + return pdata; nodata: @@ -1412,6 +1430,8 @@ static int davinci_mcasp_probe(struct platform_device *pdev) goto err; } + mcasp->ahclkx_pin = pdata->clk_input_pin; + mcasp->op_mode = pdata->op_mode; /* sanity check for tdm slots parameter */ if (mcasp->op_mode == DAVINCI_MCASP_IIS_MODE) { -- 2.1.4

2 1