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

[alsa-devel] [PATCH v4 00/10] audio timestamping evolutions
by Pierre-Louis Bossart 02 Feb '15

02 Feb '15

This series of patches was inspired by recent threads on the alsa mailing list, as well issues detected with existing and upcoming hardware: 1. there was a request from the PulseAudio community to get more information from drivers to make rewinds safer. While the conclusion was that it's nearly impossible for a driver developer to provide this information, there are however ways to assess the granularity of the hw_ptr updates using timestamping capabilities, and indirectly understand how safe rewinds might be. 2. There was also a request to add a start_at capability based either on system hr timers or a wall clock, which requires a means to expose both types of information to applications. Rather than adding new sets of timestamps, it is suggested the new start_at functionality relies on the new definition provides by these patches 3. For new hardware, there is a neeed to let low-level drivers handle timestamping instead of having the ALSA core do it. Similarly there is a need to let the low-level driver update the initial estimate for the trigger timestamp if there are delays to start a stream (eg. with USB) These patches try to provide an answer to these multiple needs by building on the work done two years ago to expose wall clock information to applications. The evolution is to let application select which audio timestamp they are interested in, track the delay and drifts between recurring measurements and get, when possible, an idea of the accuracy of the underlying hardware. A backwards compatible mode is provided in case userspace doesn't provide any timestamp selection (results based on HDAudio wallclock for playback, hw_ptr in all other cases). The first 4 patches are corrections for misses in the way the system and trigger timestamps are handled, and the last 6 provide the additional audio timestamping selection. A second batch is planned to enable hardware capabilities in a low-level drivers. A corresponding set of patches is available for alsa-lib. V2 changes: trigger_tstamp: trigger_tstamp_latched, pending redefined as bool trigger_tstamp_latched reset in snd_pcm_pre_start() audio_ts_config, report: keep separate structure but use different bitfields for in and out. use u32 instead of __u32, add comments that these structures are internal COMPAT backwards compatible mode, uses WALL_CLOCK/LINK for HDAudio playback and DEFAULT (hw_ptr) everywhere else INFO bits: reclaimed 32-bits from hw_params, renamed as info_ext moved all timestamp info to info_ext snd_pcm_status: read only 32-bit audio_tstamp_data, ignore all other fields V3/4: Addressed feedback from Jaroslav: no change to STATUS ioctl, new functionality introduced in STATUS_EXT ioctl with r/w params. bumped PCM protocol for detection on STATUS_EXT in userspace used 32-bit word for accuracy report, no mantissa/exponent packing rolled back info_ext changes, all INFO fields remain in same word Merged comments from Liam (code simplifications) fixed packing Added driver_tstamp field in case there is a delay in passing the tstamp and audio_tstamp over IPC. Pierre-Louis Bossart (10): ALSA: core: don't override timestamp unconditionally ALSA: core: allow for trigger_tstamp snapshot in .trigger ALSA: hda: read trigger_timestamp immediately after starting DMA ALSA: usb: update trigger timestamp on first non-zero URB submitted ALSA: core: selection of audio_tstamp type and accuracy reports ALSA: core: pass audio tstamp config from userspace ALSA: core: pass audio tstamp config from userspace in compat mode ALSA: core: replace .wall_clock by .get_time_info ALSA: hda: replace .wallclock by .get_time_info ALSA: bump PCM protocol to 2.0.13 Documentation/sound/alsa/timestamping.txt | 200 ++++++++++++++++++++++++++++++ include/sound/pcm.h | 67 +++++++++- include/uapi/sound/asound.h | 36 +++++- sound/core/pcm_compat.c | 36 +++++- sound/core/pcm_lib.c | 88 ++++++++----- sound/core/pcm_native.c | 58 ++++++++- sound/pci/hda/hda_controller.c | 43 +++++-- sound/usb/pcm.c | 9 ++ 8 files changed, 478 insertions(+), 59 deletions(-) create mode 100644 Documentation/sound/alsa/timestamping.txt -- 1.9.1

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[alsa-devel] [PATCH 0/3] snd-bcd2000: add audio support and control interface
by Mario Kicherer 01 Feb '15

01 Feb '15

The current version of the driver only provides a MIDI interface to read controller status and switch LEDs. These patches add audio playback and capture support as well as a control interface to switch between Phono A and Mic input.

2 7

[alsa-devel] [PATCH alsa-lib] conf: add support for USB AudioSport Quattro (tm)
by Alexander E. Patrakov 01 Feb '15

01 Feb '15

This patch resulted from discussion with redlion_ on #alsa IRC channel on Freenode. 4-channel playback now works. 4-channel capture works too, but not simultaneously with playback (hardware limitation). Alsa-info before the fix: http://www.alsa-project.org/db/?f=a3673622074b88a1abf4ccc6e7f37d0b5b72f34a Signed-off-by: Alexander E. Patrakov <patrakov(a)gmail.com> --- src/conf/cards/USB-Audio.conf | 1 + 1 file changed, 1 insertion(+) diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf index 05e5d09..4cbf85e 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -30,6 +30,7 @@ USB-Audio.pcm.surround40_type { "OmniStudio" two_stereo_devices "Quattro" two_stereo_devices "SB Audigy 2 NX" six_channels + "USB AudioSport Quattro (tm)" two_stereo_devices } # If a device does not use the first PCM device for digital data, the device -- 2.2.1

2 1

[alsa-devel] [PATCH v2 12/12] dt-bindings: Add documentation for Rockchip dw-hdmi-audio
by Yakir Yang 31 Jan '15

31 Jan '15

Required properties: - compatible: platform specific - cpu-of-node: the device node of cpu_dai Signed-off-by: Yakir Yang <ykk(a)rock-chips.com> --- Changes in v2: - remove codec-name and codec-dai-name - rename rockchip,rockchip-hdmi-audio.txt to rockchip,rockchip-dw-hdmi-audio.txt .../bindings/sound/rockchip,rockchip-dw-hdmi-audio.txt | 12 ++++++++++++ 1 file changed, 12 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rockchip-dw-hdmi-audio.txt diff --git a/Documentation/devicetree/bindings/sound/rockchip,rockchip-dw-hdmi-audio.txt b/Documentation/devicetree/bindings/sound/rockchip,rockchip-dw-hdmi-audio.txt new file mode 100644 index 0000000..5b86eed --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rockchip-dw-hdmi-audio.txt @@ -0,0 +1,12 @@ +Rockchip hdmi audio bindings + +Required properties: +- compatible: platform specific +- cpu-of-node: the device node of cpu_dai + +Example: + +sound { + compatible = "rockchip,rk3288-hdmi-audio"; + cpu-of-node = <&i2s>; +}; -- 2.1.2

3 2

[alsa-devel] [PATCH v2 10/12] ASoC: dw-hdmi-audio: add codec driver for dw hdmi audio
by Yakir Yang 31 Jan '15

31 Jan '15

codec driver creat an standard alsa device, than config audio and report jack status through some callback interfaces that dw_hdmi driver support. Signed-off-by: Yakir Yang <ykk(a)rock-chips.com> --- Changes in v2: - Update dw_hdmi audio control interfaces, and adjust jack report process sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/dw-hdmi-audio.c | 314 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/dw-hdmi-audio.h | 17 +++ 4 files changed, 337 insertions(+) create mode 100644 sound/soc/codecs/dw-hdmi-audio.c create mode 100644 sound/soc/codecs/dw-hdmi-audio.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 8349f98..b34dd12 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -75,6 +75,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C select SND_SOC_HDMI_CODEC + select SND_SOC_DW_HDMI_AUDIO select SND_SOC_PCM1681 if I2C select SND_SOC_PCM1792A if SPI_MASTER select SND_SOC_PCM3008 @@ -459,6 +460,9 @@ config SND_SOC_MAX98095 config SND_SOC_MAX9850 tristate +config SND_SOC_DW_HDMI_AUDIO + tristate + config SND_SOC_PCM1681 tristate "Texas Instruments PCM1681 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index bbdfd1e..0ebb664 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -68,6 +68,7 @@ snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o snd-soc-hdmi-codec-objs := hdmi.o +snd-soc-dw-hdmi-audio-objs := dw-hdmi-audio.o snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm1792a-codec-objs := pcm1792a.o snd-soc-pcm3008-objs := pcm3008.o @@ -249,6 +250,7 @@ obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o +obj-$(CONFIG_SND_SOC_DW_HDMI_AUDIO) += snd-soc-dw-hdmi-audio.o obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o diff --git a/sound/soc/codecs/dw-hdmi-audio.c b/sound/soc/codecs/dw-hdmi-audio.c new file mode 100644 index 0000000..ed046ce --- /dev/null +++ b/sound/soc/codecs/dw-hdmi-audio.c @@ -0,0 +1,314 @@ +/* + * dw-hdmi-codec.c + * + * DesignerWare ALSA SoC DAI driver for DW HDMI audio. + * Copyright (c) 2014, CORPORATION. All rights reserved. + * Authors: Yakir Yang <ykk(a)rock-chips.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope 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. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>.* + * + */ +#include <linux/init.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/moduleparam.h> + +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/initval.h> +#include <sound/pcm_params.h> + +#include <drm/bridge/dw_hdmi.h> +#include "dw-hdmi-audio.h" + +struct snd_dw_hdmi { + struct device *dev; + struct dw_hdmi_audio_data data; + + u8 jack_status; + struct snd_soc_jack *jack; + struct delayed_work jack_work; +}; + +/** + * dw_hdmi_jack_detect - Enable hdmi detection via the HDMI IRQ + * + * @codec: HDMI codec + * @jack: jack to report detection events on + * + * Enable HDMI detection via IRQ on the HDMI. + * + * If no jack is supplied detection will be disabled. + */ +int dw_hdmi_jack_detect(struct snd_soc_codec *codec_dai, + struct snd_soc_jack *jack) +{ + struct snd_dw_hdmi *hdmi = snd_soc_codec_get_drvdata(codec_dai); + + hdmi->jack = jack; + hdmi->jack_status = 0; + + snd_soc_jack_report(hdmi->jack, 0, SND_JACK_LINEOUT); + + queue_delayed_work(system_power_efficient_wq, &hdmi->jack_work, + msecs_to_jiffies(50)); + + return 0; +} +EXPORT_SYMBOL_GPL(dw_hdmi_jack_detect); + +static void dw_hdmi_jack_work(struct work_struct *work) +{ + struct snd_dw_hdmi *hdmi = container_of(work, struct snd_dw_hdmi, + jack_work.work); + u8 jack_status, status; + + status = hdmi->data.get_connect_status(hdmi->data.dw); + jack_status = status ? SND_JACK_LINEOUT : 0; + if (jack_status != hdmi->jack_status) { + snd_soc_jack_report(hdmi->jack, jack_status, + SND_JACK_LINEOUT); + hdmi->jack_status = jack_status; + + dev_dbg(hdmi->dev, "jack report [%d]\n", hdmi->jack_status); + } +} + +static irqreturn_t snd_dw_hdmi_irq(int irq, void *dev_id) +{ + struct snd_dw_hdmi *hdmi = dev_id; + + queue_delayed_work(system_power_efficient_wq, &hdmi->jack_work, + msecs_to_jiffies(50)); + + return IRQ_HANDLED; +} + +static int dw_hdmi_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *codec_dai) +{ + struct snd_dw_hdmi *hdmi = snd_soc_dai_get_drvdata(codec_dai); + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct hdmi_audio_fmt hdmi_fmt; + unsigned int fmt, rate, chan, width; + + fmt = rtd->dai_link->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK; + switch (fmt) { + case SND_SOC_DAIFMT_I2S: + hdmi_fmt.dai_fmt = AUDIO_DAIFMT_IIS; + break; + case SND_SOC_DAIFMT_LEFT_J: + hdmi_fmt.dai_fmt = AUDIO_DAIFMT_LEFT_J; + break; + case SND_SOC_DAIFMT_RIGHT_J: + hdmi_fmt.dai_fmt = AUDIO_DAIFMT_RIGHT_J; + break; + default: + dev_err(codec_dai->dev, "DAI format unsupported"); + return -EINVAL; + } + dev_dbg(codec_dai->dev, "[codec_dai]: dai_fmt = %d.\n", fmt); + + width = params_width(params); + switch (width) { + case 16: + case 24: + hdmi_fmt.word_length = width; + break; + default: + dev_err(codec_dai->dev, "width[%d] not support!\n", width); + return -EINVAL; + } + dev_dbg(codec_dai->dev, "[codec_dai]: word_length = %d.\n", width); + + chan = params_channels(params); + switch (chan) { + case 2: + hdmi_fmt.chan_num = AUDIO_CHANNELNUM_2; + break; + case 4: + hdmi_fmt.chan_num = AUDIO_CHANNELNUM_4; + break; + case 6: + hdmi_fmt.chan_num = AUDIO_CHANNELNUM_6; + break; + case 8: + hdmi_fmt.chan_num = AUDIO_CHANNELNUM_8; + break; + default: + dev_err(codec_dai->dev, "channel[%d] not support!\n", chan); + return -EINVAL; + } + dev_dbg(codec_dai->dev, "[codec_dai]: chan_num = %d.\n", chan); + + rate = params_rate(params); + switch (rate) { + case 32000: + case 44100: + case 48000: + case 88200: + case 96000: + case 176400: + case 192000: + hdmi_fmt.sample_rate = rate; + break; + default: + dev_err(codec_dai->dev, "rate[%d] not support!\n", rate); + return -EINVAL; + } + dev_dbg(codec_dai->dev, "[codec_dai]: sample_rate = %d.\n", rate); + + hdmi->data.config_audio(hdmi->data.dw, &hdmi_fmt); + + return 0; +} + +static int dw_hdmi_dai_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *codec_dai) +{ + struct snd_dw_hdmi *hdmi = snd_soc_dai_get_drvdata(codec_dai); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + hdmi->data.enable_clk(hdmi->data.dw); + dev_dbg(codec_dai->dev, "[codec_dai]: trigger enable.\n"); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + hdmi->data.disable_clk(hdmi->data.dw); + dev_dbg(codec_dai->dev, "[codec_dai]: trigger disable.\n"); + break; + default: + return -EINVAL; + } + + return 0; +} + +static const struct snd_soc_dapm_widget dw_hdmi_widgets[] = { + SND_SOC_DAPM_OUTPUT("TX"), +}; + +static const struct snd_soc_dapm_route dw_hdmi_routes[] = { + { "TX", NULL, "Playback" }, +}; + +static const struct snd_soc_dai_ops dw_hdmi_dai_ops = { + .hw_params = dw_hdmi_dai_hw_params, + .trigger = dw_hdmi_dai_trigger, +}; + +static struct snd_soc_dai_driver dw_hdmi_audio_dai = { + .name = "dw-hdmi-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + }, + .ops = &dw_hdmi_dai_ops, +}; + +static const struct snd_soc_codec_driver dw_hdmi_audio = { + .dapm_widgets = dw_hdmi_widgets, + .num_dapm_widgets = ARRAY_SIZE(dw_hdmi_widgets), + .dapm_routes = dw_hdmi_routes, + .num_dapm_routes = ARRAY_SIZE(dw_hdmi_routes), +}; + +static int dw_hdmi_audio_probe(struct platform_device *pdev) +{ + struct dw_hdmi_audio_data *data = pdev->dev.platform_data; + struct snd_dw_hdmi *hdmi; + int ret; + + hdmi = devm_kzalloc(&pdev->dev, sizeof(*hdmi), GFP_KERNEL); + if (!hdmi) + return -ENOMEM; + + hdmi->data = *data; + hdmi->dev = &pdev->dev; + platform_set_drvdata(pdev, hdmi); + + INIT_DELAYED_WORK(&hdmi->jack_work, dw_hdmi_jack_work); + + ret = devm_request_irq(&pdev->dev, hdmi->data.irq, snd_dw_hdmi_irq, + IRQF_SHARED, "dw-hdmi-audio", hdmi); + if (ret) { + dev_err(&pdev->dev, "request irq failed (%d)\n", ret); + goto free_hdmi_data; + } + + ret = snd_soc_register_codec(&pdev->dev, &dw_hdmi_audio, + &dw_hdmi_audio_dai, 1); + if (ret) { + dev_err(&pdev->dev, "register codec failed (%d)\n", ret); + goto free_irq; + } + + dev_info(&pdev->dev, "hdmi audio init success.\n"); + + return 0; + +free_irq: + devm_free_irq(&pdev->dev, hdmi->data.irq, hdmi); +free_hdmi_data: + devm_kfree(&pdev->dev, hdmi); + + return ret; +} + +static int dw_hdmi_audio_remove(struct platform_device *pdev) +{ + struct snd_dw_hdmi *hdmi = platform_get_drvdata(pdev); + + snd_soc_unregister_codec(&pdev->dev); + devm_free_irq(&pdev->dev, hdmi->data.irq, hdmi); + devm_kfree(&pdev->dev, hdmi); + + return 0; +} + +static const struct of_device_id dw_hdmi_audio_ids[] = { + { .compatible = "dw-hdmi-audio", }, + { } +}; + +static struct platform_driver dw_hdmi_audio_driver = { + .driver = { + .name = "dw-hdmi-audio", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(dw_hdmi_audio_ids), + }, + .probe = dw_hdmi_audio_probe, + .remove = dw_hdmi_audio_remove, +}; +module_platform_driver(dw_hdmi_audio_driver); + +MODULE_AUTHOR("Yakir Yang <ykk(a)rock-chips.com>"); +MODULE_DESCRIPTION("DW HDMI Audio ASoC Interface"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" "dw-hdmi-audio"); +MODULE_DEVICE_TABLE(of, dw_hdmi_audio_ids); diff --git a/sound/soc/codecs/dw-hdmi-audio.h b/sound/soc/codecs/dw-hdmi-audio.h new file mode 100644 index 0000000..2a7b5b4 --- /dev/null +++ b/sound/soc/codecs/dw-hdmi-audio.h @@ -0,0 +1,17 @@ +/* + * dw-hdmi-audio.h -- DW HDMI ALSA SoC Audio driver + * + * Copyright 2011-2012 DesignerWare Products + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _DW_HDMI_AUDIO_H +#define _DW_HDMI_AUDIO_H + +int dw_hdmi_jack_detect(struct snd_soc_codec *codec_dai, + struct snd_soc_jack *jack); + +#endif -- 2.1.2

2 1

[alsa-devel] [PATCH 1/5] sync asound.h with kernel version
by Pierre-Louis Bossart 31 Jan '15

31 Jan '15

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/sound/asound.h | 37 ++++++++++++++++++++++++++++++++----- 1 file changed, 32 insertions(+), 5 deletions(-) diff --git a/include/sound/asound.h b/include/sound/asound.h index 1f23cd6..3d46e9a 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -140,7 +140,7 @@ struct snd_hwdep_dsp_image { * * *****************************************************************************/ -#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 12) +#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 13) typedef unsigned long snd_pcm_uframes_t; typedef signed long snd_pcm_sframes_t; @@ -267,9 +267,17 @@ typedef int __bitwise snd_pcm_subformat_t; #define SNDRV_PCM_INFO_JOINT_DUPLEX 0x00200000 /* playback and capture stream are somewhat correlated */ #define SNDRV_PCM_INFO_SYNC_START 0x00400000 /* pcm support some kind of sync go */ #define SNDRV_PCM_INFO_NO_PERIOD_WAKEUP 0x00800000 /* period wakeup can be disabled */ -#define SNDRV_PCM_INFO_HAS_WALL_CLOCK 0x01000000 /* has audio wall clock for audio/system time sync */ +#define SNDRV_PCM_INFO_HAS_WALL_CLOCK 0x01000000 /* (Deprecated)has audio wall clock for audio/system time sync */ +#define SNDRV_PCM_INFO_HAS_LINK_ATIME 0x01000000 /* report hardware link audio time, reset on startup */ +#define SNDRV_PCM_INFO_HAS_LINK_ABSOLUTE_ATIME 0x02000000 /* report absolute hardware link audio time, not reset on startup */ +#define SNDRV_PCM_INFO_HAS_LINK_ESTIMATED_ATIME 0x04000000 /* report estimated link audio time */ +#define SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME 0x08000000 /* report synchronized audio/system time */ + +#define SNDRV_PCM_INFO_DRAIN_TRIGGER 0x40000000 /* internal kernel flag - trigger in drain */ #define SNDRV_PCM_INFO_FIFO_IN_FRAMES 0x80000000 /* internal kernel flag - FIFO size is in frames */ + + typedef int __bitwise snd_pcm_state_t; #define SNDRV_PCM_STATE_OPEN ((__force snd_pcm_state_t) 0) /* stream is open */ #define SNDRV_PCM_STATE_SETUP ((__force snd_pcm_state_t) 1) /* stream has a setup */ @@ -407,6 +415,22 @@ struct snd_pcm_channel_info { unsigned int step; /* samples distance in bits */ }; +enum { + /* + * first definition for backwards compatibility only, + * maps to wallclock/link time for HDAudio playback and DEFAULT/DMA time for everything else + */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_COMPAT = 0, + + /* timestamp definitions */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT = 1, /* DMA time, reported as per hw_ptr */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK = 2, /* link time reported by sample or wallclock counter, reset on startup */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_ABSOLUTE = 3, /* link time reported by sample or wallclock counter, not reset on startup */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_ESTIMATED = 4, /* link time estimated indirectly */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED = 5, /* link time synchronized with system time */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LAST = SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED +}; + struct snd_pcm_status { snd_pcm_state_t state; /* stream state */ struct timespec trigger_tstamp; /* time when stream was started/stopped/paused */ @@ -418,9 +442,11 @@ struct snd_pcm_status { snd_pcm_uframes_t avail_max; /* max frames available on hw since last status */ snd_pcm_uframes_t overrange; /* count of ADC (capture) overrange detections from last status */ snd_pcm_state_t suspended_state; /* suspended stream state */ - __u32 reserved_alignment; /* must be filled with zero */ - struct timespec audio_tstamp; /* from sample counter or wall clock */ - unsigned char reserved[56-sizeof(struct timespec)]; /* must be filled with zero */ + __u32 audio_tstamp_data; /* needed for 64-bit alignment, used for configs/report to/from userspace */ + struct timespec audio_tstamp; /* sample counter, wall clock, PHC or on-demand sync'ed */ + struct timespec driver_tstamp; /* useful in case reference system tstamp is reported with delay */ + __u32 audio_tstamp_accuracy; /* in ns units, only valid if indicated in audio_tstamp_data */ + unsigned char reserved[52-2*sizeof(struct timespec)]; /* must be filled with zero */ }; struct snd_pcm_mmap_status { @@ -533,6 +559,7 @@ enum { #define SNDRV_PCM_IOCTL_DELAY _IOR('A', 0x21, snd_pcm_sframes_t) #define SNDRV_PCM_IOCTL_HWSYNC _IO('A', 0x22) #define SNDRV_PCM_IOCTL_SYNC_PTR _IOWR('A', 0x23, struct snd_pcm_sync_ptr) +#define SNDRV_PCM_IOCTL_STATUS_EXT _IOWR('A', 0x24, struct snd_pcm_status) #define SNDRV_PCM_IOCTL_CHANNEL_INFO _IOR('A', 0x32, struct snd_pcm_channel_info) #define SNDRV_PCM_IOCTL_PREPARE _IO('A', 0x40) #define SNDRV_PCM_IOCTL_RESET _IO('A', 0x41) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: tlv320aic3x: Fix data delay configuration
by Peter Ujfalusi 30 Jan '15

30 Jan '15

Fix the issue introduced by: 368494093354 ASoC: tlv320aic3x: Add TDM support The CTRLC register were not receiving the correct delay configuration, which will corrupt DSP_A audio mode. Reported-by: Pavel Machek <pavel(a)ucw.cz> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, most of my boards are using DSP_B or I2S mode... This is why this is slipped. Peter sound/soc/codecs/tlv320aic3x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index a78ad361f494..07603d142923 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1046,7 +1046,7 @@ static int aic3x_prepare(struct snd_pcm_substream *substream, delay += aic3x->tdm_delay; /* Configure data delay */ - snd_soc_write(codec, AIC3X_ASD_INTF_CTRLC, aic3x->tdm_delay); + snd_soc_write(codec, AIC3X_ASD_INTF_CTRLC, delay); return 0; } -- 2.2.2

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[alsa-devel] [PATCH] ASoC: Add sysfs entries via static attribute groups
by Takashi Iwai 30 Jan '15

30 Jan '15

Instead of calling device_create_file() manually, assign the static attribute group entries at the device registration. This simplifies the code and avoids the possible races. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/soc-dapm.h | 2 +- sound/soc/soc-core.c | 40 ++++++++++++++++------------------------ sound/soc/soc-dapm.c | 16 +++++++--------- 3 files changed, 24 insertions(+), 34 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 89823cfe6f04..9fecd6752fcb 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -405,7 +405,7 @@ int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_context *dapm, struct snd_soc_dapm_update *update); /* dapm sys fs - used by the core */ -int snd_soc_dapm_sys_add(struct device *dev); +extern const struct attribute_group soc_dapm_dev_attr_group; void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm, struct dentry *parent); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2c62620abca6..386365f4a628 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -949,8 +949,6 @@ static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order) /* unregister the rtd device */ if (rtd->dev_registered) { - device_remove_file(rtd->dev, &dev_attr_pmdown_time); - device_remove_file(rtd->dev, &dev_attr_codec_reg); device_unregister(rtd->dev); rtd->dev_registered = 0; } @@ -1103,6 +1101,21 @@ err_probe: return ret; } +static struct attribute *soc_dev_attrs[] = { + &dev_attr_codec_reg.attr, + &dev_attr_pmdown_time.attr, + NULL +}; + +static const struct attribute_group soc_dev_attr_group = { + .attrs = soc_dev_attrs, +}; + +static const struct attribute_group *soc_dev_attr_groups[] = { + &soc_dev_attr_group, + &soc_dapm_dev_attr_group, +}; + static void rtd_release(struct device *dev) { kfree(dev); @@ -1120,6 +1133,7 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd, device_initialize(rtd->dev); rtd->dev->parent = rtd->card->dev; rtd->dev->release = rtd_release; + rtd->dev->groups = soc_dev_attr_groups; dev_set_name(rtd->dev, "%s", name); dev_set_drvdata(rtd->dev, rtd); mutex_init(&rtd->pcm_mutex); @@ -1136,23 +1150,6 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd, return ret; } rtd->dev_registered = 1; - - if (rtd->codec) { - /* add DAPM sysfs entries for this codec */ - ret = snd_soc_dapm_sys_add(rtd->dev); - if (ret < 0) - dev_err(rtd->dev, - "ASoC: failed to add codec dapm sysfs entries: %d\n", - ret); - - /* add codec sysfs entries */ - ret = device_create_file(rtd->dev, &dev_attr_codec_reg); - if (ret < 0) - dev_err(rtd->dev, - "ASoC: failed to add codec sysfs files: %d\n", - ret); - } - return 0; } @@ -1308,11 +1305,6 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) } #endif - ret = device_create_file(rtd->dev, &dev_attr_pmdown_time); - if (ret < 0) - dev_warn(rtd->dev, "ASoC: failed to add pmdown_time sysfs: %d\n", - ret); - if (cpu_dai->driver->compress_dai) { /*create compress_device"*/ ret = soc_new_compress(rtd, num); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c5136bb1f982..a997bb8cae93 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2127,15 +2127,14 @@ static ssize_t dapm_widget_show(struct device *dev, static DEVICE_ATTR(dapm_widget, 0444, dapm_widget_show, NULL); -int snd_soc_dapm_sys_add(struct device *dev) -{ - return device_create_file(dev, &dev_attr_dapm_widget); -} +static struct attribute *soc_dapm_dev_attrs[] = { + &dev_attr_dapm_widget.attr, + NULL +}; -static void snd_soc_dapm_sys_remove(struct device *dev) -{ - device_remove_file(dev, &dev_attr_dapm_widget); -} +const struct attribute_group soc_dapm_dev_attr_group = { + .attrs = soc_dapm_dev_attrs, +}; static void dapm_free_path(struct snd_soc_dapm_path *path) { @@ -3903,7 +3902,6 @@ void snd_soc_dapm_auto_nc_pins(struct snd_soc_card *card) */ void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm) { - snd_soc_dapm_sys_remove(dapm->dev); dapm_debugfs_cleanup(dapm); dapm_free_widgets(dapm); list_del(&dapm->list); -- 2.2.2

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[alsa-devel] [PATCH] ASoC: rt286: Add rt288 codec support
by Bard Liao 30 Jan '15

30 Jan '15

This patch adds support for rt288 codec. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt286.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 81bdd276..2eafbcf 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -34,6 +34,7 @@ #include "rt286.h" #define RT286_VENDOR_ID 0x10ec0286 +#define RT288_VENDOR_ID 0x10ec0288 struct rt286_priv { struct regmap *regmap; @@ -1210,7 +1211,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, regmap_read(rt286->regmap, RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &ret); - if (ret != RT286_VENDOR_ID) { + if (ret != RT286_VENDOR_ID && ret != RT288_VENDOR_ID) { dev_err(&i2c->dev, "Device with ID register %x is not rt286\n", ret); return -ENODEV; -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH] ASoC: rt286: set idle_bias_off to false
by Bard Liao 30 Jan '15

30 Jan '15

We do the power down process at bias level STANDBY. And do nothing at bias level OFF. As a result, the register setting is the same between bias level STANDBY and OFF. So we don't need to set idle_bias_off true. Signed-off-by: Jie Yang <yang.jie(a)intel.com> Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt286.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 847cc4b..81bdd276 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1144,7 +1144,7 @@ static struct snd_soc_codec_driver soc_codec_dev_rt286 = { .suspend = rt286_suspend, .resume = rt286_resume, .set_bias_level = rt286_set_bias_level, - .idle_bias_off = true, + .idle_bias_off = false, .controls = rt286_snd_controls, .num_controls = ARRAY_SIZE(rt286_snd_controls), .dapm_widgets = rt286_dapm_widgets, -- 1.8.1.1.439.g50a6b54

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