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/linux/Documentation/sound/cards/
H A Dcmipci.rst2 Brief Notes on C-Media 8338/8738/8768/8770 Driver
8 Front/Rear Multi-channel Playback
9 ---------------------------------
13 DACs, both streams are handled independently unlike the 4/6ch multi-
18 (hw:0,1) is assigned to the second DAC for rear playback.
22 - The first DAC supports U8 and S16LE formats, while the second DAC
23 supports only S16LE.
24 - The second DAC supports only two channel stereo.
26 Please note that the CM8x38 DAC doesn't support continuous playback
27 rate but only fixed rates: 5512, 8000, 11025, 16000, 22050, 32000,
[all …]
H A Demu-mixer.rst2 E-MU Digital Audio System mixer / default DSP code
5 This document covers the E-MU 0404/1010/1212/1616/1820 PCI/PCI-e/CardBus
9 alternative front-end geared towards semi-professional studio recording.
11 This document is based on audigy-mixer.rst.
20 This is the case on more modern motherboards, where the PCI bus is only a
22 In particular, I got recording glitches during simultaneous playback on an
33 This driver supports only 16-bit 44.1/48 kHz operation. The multi-channel
34 device (see emu10k1-jack.rst) additionally supports 24-bit capture.
37 <https://github.com/ossilator/linux/tree/ossis-emu10k1>`_.
38 Its multi-channel device supports 24-bit for both playback and capture,
[all …]
H A Daudigy-mixer.rst5 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
47 name='PCM Front Playback Volume',index=0
48 ----------------------------------------
49 This control is used to attenuate samples from left and right front PCM FX-bus
51 samples for 5.1 playback. The result samples are forwarded to the front speakers.
53 name='PCM Surround Playback Volume',index=0
54 -------------------------------------------
[all …]
/linux/Documentation/sound/designs/
H A Dtimestamping.rst7 - Trigger_tstamp is the system time snapshot taken when the .trigger
10 capabilities or conversely may only be able to provide a correct
11 estimate with a delay. In the latter two cases, the low-level driver
17 - tstamp is the current system timestamp updated during the last
19 The difference (tstamp - trigger_tstamp) defines the elapsed time.
29 - ``avail`` reports how much can be written in the ring buffer
30 - ``delay`` reports the time it will take to hear a new sample after all
43 ascii-art, this could be represented as follows (for the playback
47 --------------------------------------------------------------> time
53 |< codec delay >|<--hw delay-->|<queued samples>|<---avail->|
[all …]
H A Doss-emulation.rst2 Notes on Kernel OSS-Emulation
13 as add-on kernel modules, snd-pcm-oss, snd-mixer-oss and snd-seq-oss.
18 is called. The alias is defined ``sound-service-x-y``, where x and y are
22 Only necessary step for auto-loading of OSS modules is to define the
25 alias sound-slot-0 snd-emu10k1
27 As the second card, define ``sound-slot-1`` as well.
29 ``alias sound-slot-0 snd-card-0`` doesn't work any more like the old
37 Please note that the devices listed in this proc file appear only
38 after the corresponding OSS-emulation module is loaded. Don't worry
71 /dev/dsp2, but only /dev/dsp0 and /dev/adsp0.
[all …]
H A Dcontrol-names.rst8 ---------------
16 Playback one direction
18 Bypass Playback one direction
68 Headset Mic mic part of combined headset jack - 4-pin
70 Headphone Mic mic part of either/or - 3-pin headphone or mic
71 Line input only, use "Line Out" for output
79 Analog Loopback D/A -> A/D loopback
80 Digital Loopback playback -> capture loopback -
91 SPDIF output only
98 -----------------------
[all …]
/linux/sound/soc/codecs/
H A Dad1980.c1 // SPDX-License-Identifier: GPL-2.0-or-later
3 * ad1980.c -- ALSA Soc AD1980 codec support
112 SOC_DOUBLE("Master Playback Volume", AC97_MASTER, 8, 0, 31, 1),
113 SOC_SINGLE("Master Playback Switch", AC97_MASTER, 15, 1, 1),
115 SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1),
116 SOC_SINGLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 1, 1),
118 SOC_DOUBLE("PCM Playback Volume", AC97_PCM, 8, 0, 31, 1),
119 SOC_SINGLE("PCM Playback Switch", AC97_PCM, 15, 1, 1),
124 SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 1),
125 SOC_SINGLE("Mono Playback Switch", AC97_MASTER_MONO, 15, 1, 1),
[all …]
H A Dadau1761.c1 // SPDX-License-Identifier: GPL-2.0-or-later
5 * Copyright 2011-2013 Analog Devices Inc.
6 * Author: Lars-Peter Clausen <lars@metafoo.de>
125 static const DECLARE_TLV_DB_SCALE(adau1761_sing_in_tlv, -1500, 300, 1);
126 static const DECLARE_TLV_DB_SCALE(adau1761_diff_in_tlv, -1200, 75, 0);
127 static const DECLARE_TLV_DB_SCALE(adau1761_out_tlv, -5700, 100, 0);
128 static const DECLARE_TLV_DB_SCALE(adau1761_sidetone_tlv, -1800, 300, 1);
129 static const DECLARE_TLV_DB_SCALE(adau1761_boost_tlv, -600, 600, 1);
130 static const DECLARE_TLV_DB_SCALE(adau1761_pga_boost_tlv, -2000, 2000, 1);
132 static const DECLARE_TLV_DB_SCALE(adau1761_alc_max_gain_tlv, -1200, 600, 0);
[all …]
H A Dak4554.c1 // SPDX-License-Identifier: GPL-2.0
15 * ak4554 playback format is SND_SOC_DAIFMT_RIGHT_J,
22 * CPU-DAI1 (plaback only fmt = RIGHT_J) --+-- ak4554
24 * CPU-DAI2 (capture only fmt = LEFT_J) ---+
39 { "AOUTL", NULL, "Playback" },
40 { "AOUTR", NULL, "Playback" },
44 .name = "ak4554-hifi",
45 .playback = {
46 .stream_name = "Playback",
74 return devm_snd_soc_register_component(&pdev->dev, in ak4554_soc_probe()
[all …]
H A Dtwl4030.c1 // SPDX-License-Identifier: GPL-2.0-only
13 #include <linux/mfd/twl4030-audio.h>
64 u8 ctl_cache[TWL4030_REG_PRECKR_CTL - TWL4030_REG_EAR_CTL + 1];
76 twl4030->ctl_cache[i - TWL4030_REG_EAR_CTL] = byte; in tw4030_init_ctl_cache()
86 return -EIO; in twl4030_read()
95 value = twl4030->ctl_cache[reg - TWL4030_REG_EAR_CTL]; in twl4030_read()
113 if (twl4030->earpiece_enabled) in twl4030_can_write_to_chip()
117 if (twl4030->predrivel_enabled) in twl4030_can_write_to_chip()
121 if (twl4030->predriver_enabled) in twl4030_can_write_to_chip()
125 if (twl4030->carkitl_enabled) in twl4030_can_write_to_chip()
[all …]
H A Dak4613.c1 // SPDX-License-Identifier: GPL-2.0
3 // ak4613.c -- Asahi Kasei ALSA Soc Audio driver
14 * +-------+
16 * SDTO1 <-| |
18 * SDTI1 ->| |
19 * SDTI2 ->| |
20 * SDTI3 ->| |
21 * +-------+
23 * +---+
42 * Playback 2ch : SDTI1
[all …]
H A Dwm9705.c1 // SPDX-License-Identifier: GPL-2.0-only
3 * wm9705.c -- ALSA Soc WM9705 codec support
76 SOC_DOUBLE("Master Playback Volume", AC97_MASTER, 8, 0, 31, 1),
77 SOC_SINGLE("Master Playback Switch", AC97_MASTER, 15, 1, 1),
78 SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1),
79 SOC_SINGLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 1, 1),
80 SOC_DOUBLE("PCM Playback Volume", AC97_PCM, 8, 0, 31, 1),
81 SOC_SINGLE("PCM Playback Switch", AC97_PCM, 15, 1, 1),
82 SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 1),
83 SOC_SINGLE("Mono Playback Switch", AC97_MASTER_MONO, 15, 1, 1),
[all …]
/linux/sound/pci/ice1712/
H A Dpsc724.c1 // SPDX-License-Identifier: GPL-2.0-or-later
7 * Copyright (c) 2012 Ondrej Zary <linux@rainbow-software.org>
34 * VT1722 (Envy24GT) - 6 outputs, 4 inputs (only 2 used), 24-bit/96kHz
42 * AC-Link configuration ICE_EEP2_ACLINK=0x80
60 * 2-channel DAC used for main output and stereo ADC (with 10-channel MUX)
63 * MODE (pin16) -- GND
64 * CE (pin17) -- GND I2C mode (address=0x34)
65 * DI (pin18) -- SDA (VT1722 pin70)
66 * CL (pin19) -- SCLK (VT1722 pin71)
69 * 6-channel DAC used for rear & center/LFE outputs (only 4 channels used)
[all …]
/linux/sound/soc/renesas/
H A Drz-ssi.c1 // SPDX-License-Identifier: GPL-2.0
3 // Renesas RZ/G2L ASoC Serial Sound Interface (SSIF-2) Driver
75 #define SSI_RATES SNDRV_PCM_RATE_8000_48000 /* 8k-48kHz */
117 * The SSI supports full-duplex transmission and reception.
120 * So it is better to use as half-duplex (playing and recording
123 struct rz_ssi_stream playback; member
148 writel(data, (priv->base + reg)); in rz_ssi_reg_writel()
153 return readl(priv->bas in rz_ssi_reg_readl()
[all...]
/linux/sound/soc/fsl/
H A Dmpc5200_dma.c1 // SPDX-License-Identifier: GPL-2.0-only
10 #include <linux/dma-mapping.h>
33 struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; in psc_dma_status_irq()
36 isr = in_be16(&regs->mpc52xx_psc_isr); in psc_dma_status_irq()
38 /* Playback underrun error */ in psc_dma_status_irq()
39 if (psc_dma->playback.active && (isr & MPC52xx_PSC_IMR_TXEMP)) in psc_dma_status_irq()
40 psc_dma->stats.underrun_count++; in psc_dma_status_irq()
43 if (psc_dma->capture.active && (isr & MPC52xx_PSC_IMR_ORERR)) in psc_dma_status_irq()
44 psc_dma->stats.overrun_count++; in psc_dma_status_irq()
46 out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT); in psc_dma_status_irq()
[all …]
/linux/include/sound/
H A Dsoc_sdw_utils.h1 /* SPDX-License-Identifier: GPL-2.0-only */
13 #include <sound/soc-acpi.h>
25 * - 0 - No speaker output
26 * - SOC_SDW_CODEC_SPKR - CODE
[all...]
/linux/sound/pci/emu10k1/
H A Dp16v.h1 /* SPDX-License-Identifier: GPL-2.0-or-later */
3 * Copyright (c) by James Courtier-Dutton <James@superbug.demon.co.uk>
11 /* Audigy2 P16V pointer-offset register set, accessed through the PTR2 and DATA2 registers …
25 #define PLAYBACK_LIST_SIZE 0x01 /* Size of list in bytes << 16. E.g. 8 periods -> 0x00380000 */
28 #define PLAYBACK_DMA_ADDR 0x04 /* Playback DMA address */
29 #define PLAYBACK_PERIOD_SIZE 0x05 /* Playback period size. win2000 uses 0x04000000 */
30 #define PLAYBACK_POINTER 0x06 /* Playback period pointer. Used with PLAYBACK_LIST_PTR to determine…
31 #define PLAYBACK_FIFO_END_ADDRESS 0x07 /* Playback FIFO end address */
32 #define PLAYBACK_FIFO_POINTER 0x08 /* Playback FIFO pointer and number of valid sound samples in c…
57 * [9:8] Playback input 0 channel select. 0 = Play output 0.
[all …]
/linux/Documentation/devicetree/bindings/sound/
H A Dfsl,ssi.yaml1 # SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
3 ---
5 $schema: http://devicetree.org/meta-schemas/core.yaml#
10 - Shengjiu Wang <shengjiu.wang@nxp.com>
13 Notes on fsl,playback-dma and fsl,capture-dma
14 On SOCs that have an SSI, specific DMA channels are hard-wired for playback
16 playback and DMA channel 1 for capture. SSI2 must use DMA channel 2 for
17 playback and DMA channel 3 for capture. The developer can choose which
18 DMA controller to use, but the channels themselves are hard-wired. The
22 "fsl,playback-dma" and "fsl,capture-dma" must be marked as compatible with
[all …]
/linux/sound/pci/ca0106/
H A Dca0106.h1 /* SPDX-License-Identifier: GPL-2.0-or-later */
3 * Copyright (c) 2004 James Courtier-Dutton <James@superbug.demon.co.uk>
24 …* Enable playback 48000 and 96000 rates. (Rates other that these do not work, even with "plug:f…
28 …* Correct interrupt timing. interrupt at end of period, instead of in the middle of a playback …
34 * playback periods_min=2, periods_max=8
36 * playback hw constraints require period_size = n * 64 bytes.
50 * Implement support for Line-in capture on SB Live 24bit.
73 #define IPR_MIDI_RX_B 0x00020000 /* MIDI UART-B Receive buffer non-empty */
74 #define IPR_MIDI_TX_B 0x00010000 /* MIDI UART-B Transmit buffer empty */
87 #define IPR_MIDI_RX_A 0x00000004 /* MIDI UART-A Receive buffer non-empty */
[all …]
/linux/sound/isa/sb/
H A Dsb_mixer.c1 // SPDX-License-Identifier: GPL-2.0-or-later
24 dev_dbg(chip->card->dev, "mixer_write 0x%x 0x%x\n", reg, data); in snd_sbmixer_write()
37 dev_dbg(chip->card->dev, "mixer_read 0x%x 0x%x\n", reg, result); in snd_sbmixer_read()
48 int mask = (kcontrol->private_valu in snd_sbmixer_info_single()
[all...]
/linux/sound/pci/
H A Dazt3328.h1 /* SPDX-License-Identifier: GPL-2.0 */
5 /* "PU" == "power-up value", as tested on PCI168 PCI rev. 10
9 /* (only 0x70 of 0x80 bytes saved/restored by Windows driver) */
15 * from 0x00 (playback codec), from 0x20 (recording codec)
27 * 0x0001 is the only bit that's able to start the DMA counter */
30 * both 0x0002 and 0x0004 set in playback setup. */
68 * have any hard facts, only rough measurements.
85 …REQ_SUSPECTED_66200 0x06 | SOUNDFORMAT_XTAL2 /* 66200 (13240 * 5); 64000 may have been nicer :-\ */
111 #define IDX_IO_TIMER_VALUE 0x60 /* found this timer area by pure luck :-) */
122 /* some IRQ bit in here might also be used to signal a power-management timer
[all …]
/linux/sound/soc/sof/amd/
H A Dacp70.c1 // SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause)
18 #include "../sof-audio.h"
20 #include "acp-dsp-offset.h"
31 .name = "acp-sof-hs",
32 .playback = {
45 /* Supporting only stereo for I2S HS controller capture */
55 .name = "acp-sof-bt",
56 .playback = {
69 /* Supporting only stereo for I2S BT controller capture */
79 .name = "acp-sof-sp",
[all …]
H A Dacp63.c1 // SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause)
18 #include "../sof-audio.h"
20 #include "acp-dsp-offset.h"
31 .name = "acp-sof-hs",
32 .playback = {
45 /* Supporting only stereo for I2S HS controller capture */
55 .name = "acp-sof-bt",
56 .playback = {
69 /* Supporting only stereo for I2S BT controller capture */
79 .name = "acp-sof-sp",
[all …]
H A Drembrandt.c1 // SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause)
18 #include "../sof-audio.h"
20 #include "acp-dsp-offset.h"
31 .name = "acp-sof-hs",
32 .playback = {
45 /* Supporting only stereo for I2S HS controller capture */
55 .name = "acp-sof-bt",
56 .playback = {
69 /* Supporting only stereo for I2S BT controller capture */
79 .name = "acp-sof-sp",
[all …]
/linux/sound/usb/line6/
H A Dpcm.h1 /* SPDX-License-Identifier: GPL-2.0-only */
5 * Copyright (C) 2004-2010 Markus Grabner (line6@grabner-graz.at)
23 with only one frame per packet.
38 (line6pcm->pcm->streams[stream].substream)
45 *) PCM playback and capture via ALSA
49 capture and playback stream, which must be shared between these
53 We define two bit flags, "opened" and "running", for each playback
56 LINE6_STREAM_PCM = ALSA PCM playback or capture
149 /* Capture and playback streams */
163 /* PCM playback volume (left and right) */

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