| /linux/Documentation/userspace-api/media/v4l/ |
| H A D | vidioc-g-tuner.rst | 1 .. SPDX-License-Identifier: GFDL-1.1-no-invariants-or-later
13 VIDIOC_G_TUNER - VIDIOC_S_TUNER - Get or set tuner attributes
52 Since this is a write-only ioctl, it does not return the actually
68 .. flat-table:: struct v4l2_tuner
69 :header-rows: 0
70 :stub-columns: 0
72 * - __u32
73 - ``index``
74 - :cspan:`1` Identifies the tuner, set by the application.
75 * - __u8
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| H A D | vidioc-g-modulator.rst | 1 .. SPDX-License-Identifier: GFDL-1.1-no-invariants-or-later
13 VIDIOC_G_MODULATOR - VIDIOC_S_MODULATOR - Get or set modulator attributes
52 this is a write-only ioctl, it does not return the actual audio
67 .. flat-table:: struct v4l2_modulator
68 :header-rows: 0
69 :stub-columns: 0
72 * - __u32
73 - ``index``
74 - Identifies the modulator, set by the application.
75 * - __u8
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| /linux/Documentation/sound/cards/ |
| H A D | audigy-mixer.rst | 5 This is based on sb-live-mixer.rst.
20 functionality. Only the default built-in code in the ALSA driver is described
34 one-way three wire serial bus for digital sound by Philips Semiconductors
42 FX-bus
48 ----------------------------------------
49 This control is used to attenuate samples from left and right front PCM FX-bus
54 -------------------------------------------
55 This control is used to attenuate samples from left and right surround PCM FX-bus
61 ---------------------------------------
62 This control is used to attenuate samples from left and right side PCM FX-bus
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| H A D | sb-live-mixer.rst | 19 (index 0) for a given card) allows to forward 48kHz, stereo, 16-bit
22 to 8 raw PCM devices operating at 48kHz, 16-bit little endian. It would
24 but the conversion routines exist only for stereo (2-channel streams)
34 functionality. Only the default built-in code in the ALSA driver is described
48 one-way three wire serial bus for digital sound by Philips Semiconductors
56 FX-bus
63 ---------------------------------------
64 This control is used to attenuate samples from left and right PCM FX-bus
69 ------------------------------------------------
70 This control is used to attenuate samples from left and right PCM FX-bus
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| H A D | mixart.rst | 13 The miXart8AES/EBU is the same with a add-on card that offers further
15 Furthermore the add-on card offers external clock synchronisation
23 Use the mixartloader that can be found in the alsa-tools package.
35 -------
37 Sample rates : 8000 - 48000 Hz continuously
40 --------
44 Mono files will be played on the left and right channel. Each channel
48 -------
53 -----
56 <PCM 0-3> and <PCM Capture>
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| /linux/sound/soc/codecs/ |
| H A D | Kconfig | 1 # SPDX-License-Identifier: GPL-2.0-only
5 # setting - SPI can't be modular so that case doesn't need to be covered.
498 tristate "Analog Devices AU1761 CODEC - I2C"
504 tristate "Analog Devices AU1761 CODEC - SPI"
541 tristate "Analog Devices ADAU7002 Stereo PDM-to-I2S/TDM Converter"
547 tristate "Analog Devices ADAU7118 8 Channel PD
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| H A D | rt5640.c | 1 // SPDX-License-Identifier: GPL-2.0-only
3 * rt5640.c -- RT5640/RT5639 ALSA SoC audio codec driver
27 #include <sound/soc-dapm.h>
340 static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0);
341 static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -6562, 0);
342 static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0);
343 static const DECLARE_TLV_DB_MINMAX(adc_vol_tlv, -1762, 3000);
382 SOC_DOUBLE("Speaker Channel Switch", RT5640_SPK_VOL,
387 SOC_DOUBLE("HP Channel Switch", RT5640_HP_VOL,
394 SOC_DOUBLE("OUT Channel Switch", RT5640_OUTPUT,
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| H A D | wm8974.c | 1 // SPDX-License-Identifier: GPL-2.0-only
3 * wm8974.c -- WM8974 ALSA Soc Audio driver
5 * Copyright 2006-2009 Wolfson Microelectronics PLC.
55 static const char *wm8974_companding[] = {"Off", "NC", "u-law", "A-law" };
91 static const DECLARE_TLV_DB_SCALE(digital_tlv, -12750, 50, 1);
92 static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
93 static const DECLARE_TLV_DB_SCALE(inpga_tlv, -1200, 75, 0);
94 static const DECLARE_TLV_DB_SCALE(spk_tlv, -5700, 100, 0);
103 SOC_ENUM("Playback De-emphasis", wm8974_enum[2]),
166 SOC_SINGLE("Mono Playback Switch", WM8974_MONOMIX, 6, 1, 1),
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| /linux/sound/hda/common/ |
| H A D | hda_local.h | 1 /* SPDX-License-Identifier: GPL-2.0-or-later */
17 * snd_hda_ctl_add() takes the lower-bit subdev value as a valid NID.
33 /* mono volume with index (index=0,1,...) (channel=1,2) */
34 #define HDA_CODEC_VOLUME_MONO_IDX(xname, xcidx, nid, channel, xindex, dir, flags) \ argument
44 .private_value = HDA_COMPOSE_AMP_VAL(nid, channel, xindex, dir) | flags }
48 /* mono volume */
49 #define HDA_CODEC_VOLUME_MONO(xname, nid, channel, xindex, direction) \ argument
50 HDA_CODEC_VOLUME_MONO_IDX(xname, 0, nid, channel, xindex, direction, 0)
58 /* mono mute switch with index (index=0,1,...) (channel=1,2) */
59 #define HDA_CODEC_MUTE_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \ argument
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| /linux/Documentation/sound/designs/ |
| H A D | control-names.rst | 8 ---------------
9 Syntax: [LOCATION] SOURCE [CHANNEL] [DIRECTION] FUNCTION
30 CHANNEL section in Standard Syntax
33 <nothing> channel independent, or applies to all channels
37 Center center channel
38 LFE LFE channel
55 Master Mono
68 Headset Mic mic part of combined headset jack - 4-pin
70 Headphone Mic mic part of either/or - 3-pin headphone or mic
79 Analog Loopback D/A -> A/D loopback
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| /linux/drivers/media/i2c/ |
| H A D | tvaudio.c | 14 * Copyright(c) 2005-2008 Mauro Carvalho Chehab
15 * - Some cleanups, code fixes, etc
16 * - Convert it to V4L2 API
21 * debug - set to 1 if you'd like to see debug messages
40 #include <media/v4l2-device.h>
41 #include <media/v4l2-ctrls.h>
43 /* ---------------------------------------------------------------------- */
53 #define UNSET (-1U)
55 /* ---------------------------------------------------------------------- */
97 /* functions to convert the values (v4l -> chip) */
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| H A D | msp3400-kthreads.c | 1 // SPDX-License-Identifier: GPL-2.0-or-later
5 * (c) 1997-2001 Gerd Knorr <kraxel@bytesex.org>
14 #include <media/v4l2-common.h>
15 #include <media/drv-intf/msp3400.h>
18 #include "msp3400-driver.h"
31 "4.5/4.72 M Dual FM-Stereo", V4L2_STD_MN },
33 "5.5/5.74 B/G Dual FM-Stereo", V4L2_STD_BG },
35 "6.5/6.25 D/K1 Dual FM-Stereo", V4L2_STD_DK },
37 "6.5/6.74 D/K2 Dual FM-Stereo", V4L2_STD_DK },
39 "6.5 D/K FM-Mono (HDEV3)", V4L2_STD_DK },
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| /linux/Documentation/devicetree/bindings/sound/ |
| H A D | richtek,rt9123.yaml | 1 # SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
3 ---
5 $schema: http://devicetree.org/meta-schemas/core.yaml#
10 - ChiYuan Huang <cy_huang@richtek.com>
13 RT9123 is a 3.2W mono Class-D audio amplifier that features high efficiency
14 and performance with ultra-low quiescent current. The digital audio interface
15 support various formats, including I2S, left-justified, right-justified, and
18 RTQ9124 is an ultra-low output noise, digital input, mono-channel Class-D
20 both DC and AC load diagnostics, as well as real-time load monitoring to
25 - $ref: dai-common.yaml#
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| H A D | wlf,arizona.yaml | 1 # SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
3 ---
5 $schema: http://devicetree.org/meta-schemas/core.yaml#
10 - patches@opensource.cirrus.com
20 - $ref: dai-common.yaml#
23 '#sound-dai-cells':
31 signals. Valid values are 0 (Differential), 1 (Single-ended) and
38 $ref: /schemas/types.yaml#/definitions/uint32-array
46 wlf,out-mono:
48 A list of boolean values indicating whether each output is mono
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| H A D | adi,max98388.yaml | 1 # SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
3 ---
5 $schema: http://devicetree.org/meta-schemas/core.yaml#
10 - Ryan Lee <ryans.lee@analog.com>
13 The MAX98388 is a mono Class-D speaker amplifier with I/V feedback.
18 - $ref: dai-common.yaml#
23 - adi,max98388
28 '#sound-dai-cells':
31 adi,vmon-slot-no:
38 adi,imon-slot-no:
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| H A D | st,sta32x.txt | 7 - compatible: "st,sta32x"
8 - reg: the I2C address of the device for I2C
9 - reset-gpios: a GPIO spec for the reset pin. If specified, it will be
12 - power-down-gpios: a GPIO spec for the power down pin. If specified,
16 - Vdda-supply: regulator spec, providing 3.3V
17 - Vdd3-supply: regulator spec, providing 3.3V
18 - Vcc-supply: regulator spec, providing 5V - 26V
22 - clocks, clock-names: Clock specifier for XTI input clock.
24 and disabled when it is removed. The 'clock-names' must be set to 'xti'.
26 - st,output-conf: number, Selects the output configuration:
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| H A D | cs35l34.txt | 5 - compatible : "cirrus,cs35l34"
7 - reg : the I2C address of the device for I2C.
9 - VA-supply, VP-supply : power supplies for the device,
13 - cirrus,boost-vtge-millivolt : Boost Voltage Value. Configures the boost
17 - cirrus,boost-nanohenry: Inductor value for boost converter. The value is
22 - reset-gpios: GPIO used to reset the amplifier.
24 - interrupts : IRQ line info CS35L34.
25 (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt
28 - cirrus,boost-peak-milliamp : Boost converter peak current limit in mA. The
32 - cirrus,i2s-sdinloc : ADSP SDIN I2S channel location. Indicates whether the
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| /linux/sound/pci/echoaudio/ |
| H A D | echoaudio_dsp.h | 3 Copyright Echo Digital Audio Corporation (c) 1998 - 2004
21 Foundation, Inc., 59 Temple Place - Suite 330, Boston,
22 MA 02111-1307, USA.
26 Translation from C++ and adaptation for use in ALSA-Driver
41 /**** Echo24: Gina24, Layla24, Mona, Mia, Mia-midi ****/
81 * These are the offsets for the memory-mapped DSP registers; the DSP base
133 #define MIDI_IN_SKIP_DATA (-1)
136 /*----------------------------------------------------------------------------
151 -Set the clock select bits in the control register to 0xe (see the #define
154 -Set double-speed mode if you want to use sample rates above 50 kHz
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| /linux/include/sound/ |
| H A D | wavefront.h | 1 /* SPDX-License-Identifier: GPL-2.0-or-later */
8 * Copyright (c) by Paul Barton-Davis <pbd@op.net>
19 /* Pseudo-commands not part of the WaveFront command set.
103 /* OR-values for MIDI status bits */
219 u8 mono:1; member
315 /* channel constants */
328 #define WF_SAMPLE_IS_8BIT(smpl) ((smpl)->SampleResolution&2)
334 never been copied (just mmap-ed into user space straight from the
335 disk), it would be nice to allow handling of multi-channel sample
336 data without forcing user-level extraction of the relevant bytes.
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| /linux/sound/pci/ |
| H A D | es1938.c | 1 // SPDX-License-Identifier: GPL-2.0-or-later
3 * Driver for ESS Solo-1 (ES1938, ES1946, ES1969) soundcard
7 * Abramo Bagnara <abramo@alsa-project.org>,
18 - Capture data is written unaligned starting from dma_base + 1 so I need to
20 - After several cycle of the following:
21 while : ; do arecord -d
609 int u, is8, mono; snd_es1938_capture_prepare() local
657 int u, is8, mono; snd_es1938_playback1_prepare() local
694 int u, is8, mono; snd_es1938_playback2_prepare() local
824 snd_es1938_capture_copy(struct snd_pcm_substream * substream,int channel,unsigned long pos,struct iov_iter * dst,unsigned long count) snd_es1938_capture_copy() argument
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| H A D | ad1889.c | 1 // SPDX-License-Identifier: GPL-2.0-only
5 * on the HP PA-RISC [BCJ]-xxx0 workstations.
7 * Copyright (C) 2004-2005, Kyle McMartin <kyle@parisc-linux.org>
8 * Copyright (C) 2005, Thibaut Varene <varenet@parisc-linux.org>
25 #include <linux/dma-mappin
182 ad1889_channel_reset(struct snd_ad1889 * chip,unsigned int channel) ad1889_channel_reset() argument
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| /linux/sound/usb/ |
| H A D | mixer.c | 1 // SPDX-License-Identifier: GPL-2.0-or-later
17 * - support for UAC2 effect units
18 * - support for graphical equalizers
19 * - RANGE and MEM set commands (UAC2)
20 * - RANGE and MEM interrupt dispatchers (UAC2)
21 * - audi
424 get_cur_mix_raw(struct usb_mixer_elem_info * cval,int channel,int * value) get_cur_mix_raw() argument
432 snd_usb_get_cur_mix_value(struct usb_mixer_elem_info * cval,int channel,int index,int * value) snd_usb_get_cur_mix_value() argument
519 snd_usb_set_cur_mix_value(struct usb_mixer_elem_info * cval,int channel,int index,int value) snd_usb_set_cur_mix_value() argument
3321 delegate_notify(struct usb_mixer_interface * mixer,int unitid,u8 * control,u8 * channel) delegate_notify() argument
3417 __u8 channel = value & 0xff; snd_usb_mixer_interrupt_v2() local
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| /linux/sound/pci/asihpi/ |
| H A D | hpi.h | 1 /* SPDX-License-Identifier: GPL-2.0-only */
5 Copyright (C) 1997-2011 AudioScience Inc. <support@audioscience.com>
14 The HPI is a low-level hardware abstraction layer to all
17 (C) Copyright AudioScience Inc. 1998-2010
37 /** 8-bit unsigned PCM. Windows equivalent is WAVE_FORMAT_PCM. */
39 /** 16-bit signed PCM. Windows equivalent is WAVE_FORMAT_PCM. */
41 /** MPEG-1 Layer-1. */
43 /** MPEG-1 Layer-2.
52 <td><p><b>Mono</b></p>
53 <td><p><b>Stereo,<br>Joint Stereo or<br>Dual Channel</b></p>
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| /linux/sound/pci/emu10k1/ |
| H A D | p17v.h | 1 /* SPDX-License-Identifier: GPL-2.0-or-later */
3 * Copyright (c) by James Courtier-Dutton <James@superbug.demon.co.uk>
8 /* Audigy2Value Tina (P17V) pointer-offset register set, */
12 /* 00 - 07: Not used */
16 /* 09 - 12: Not used */
20 /* 14 - 17: Not used */
21 #define P17V_PB_CHN_SEL 0x18 /* P17v playback channel select */
24 /* 1b - 1f: Not used */
25 /* 20 - 2f: Not used */
26 /* 30 - 3b: Not used */
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| /linux/drivers/media/radio/si470x/ |
| H A D | radio-si470x-common.c | 1 // SPDX-License-Identifier: GPL-2.0-or-later
3 * drivers/media/radio/si470x/radio-si470x-common.c
14 * 2008-01-12 Tobias Lorenz <tobias.lorenz@gmx.net>
16 * - First working version
17 * 2008-01-13 Tobias Lorenz <tobias.lorenz@gmx.net>
19 * - Improved error handling, every function now returns errno
20 * - Improved multi user access (start/mute/stop)
21 * - Channel doesn't get lost anymore after start/mute/stop
22 * - RDS support added (polling mode via interrupt EP 1)
23 * - marked default module parameters with *value*
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