/linux/Documentation/sound/cards/ |
H A D | cmipci.rst | 2 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 …]
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H A D | emu-mixer.rst | 2 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 …]
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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 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 …]
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/linux/Documentation/sound/designs/ |
H A D | timestamping.rst | 7 - 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 …]
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H A D | oss-emulation.rst | 2 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 …]
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H A D | control-names.rst | 8 --------------- 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 …]
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/linux/sound/soc/codecs/ |
H A D | ad1980.c | 1 // 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 …]
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H A D | adau1761.c | 1 // 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 …]
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H A D | ak4554.c | 1 // 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 …]
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H A D | wm8350.c | 1 // SPDX-License-Identifier: GPL-2.0-only 3 * wm8350.c -- WM8350 ALSA SoC audio driver 5 * Copyright (C) 2007-12 Wolfson Microelectronics PLC. 36 /* We only include the analogue supplies here; the digital supplies 77 struct wm8350_output *out1 = &wm8350_data->out1; in wm8350_out1_ramp_step() 78 struct wm8350 *wm8350 = wm8350_data->wm8350; in wm8350_out1_ramp_step() 86 if (out1->ramp == WM8350_RAMP_UP) { in wm8350_out1_ramp_step() 88 if (val < out1->left_vol) { in wm8350_out1_ramp_step() 95 } else if (out1->ramp == WM8350_RAMP_DOWN) { in wm8350_out1_ramp_step() 98 val--; in wm8350_out1_ramp_step() [all …]
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H A D | ak4613.c | 1 // 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 …]
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H A D | wm9705.c | 1 // 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 …]
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H A D | wm8770.c | 1 // SPDX-License-Identifier: GPL-2.0-only 3 * wm8770.c -- WM8770 ALSA SoC Audio driver 105 regcache_mark_dirty(wm8770->regmap); \ 114 static const DECLARE_TLV_DB_SCALE(adc_tlv, -1200, 100, 0); 115 static const DECLARE_TLV_DB_SCALE(dac_dig_tlv, -12750, 50, 1); 116 static const DECLARE_TLV_DB_SCALE(dac_alg_tlv, -12700, 100, 1); 133 /* global DAC playback controls */ 134 SOC_SINGLE_TLV("DAC Playback Volume", WM8770_MSDIGVOL, 0, 255, 0, 136 SOC_SINGLE("DAC Playback Switch", WM8770_DACMUTE, 4, 1, 1), 137 SOC_SINGLE("DAC Playback ZC Switch", WM8770_DACCTRL1, 0, 1, 0), [all …]
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/linux/sound/pci/ice1712/ |
H A D | psc724.c | 1 // 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 …]
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/linux/sound/soc/fsl/ |
H A D | mpc5200_dma.c | 1 // 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(®s->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(®s->command, MPC52xx_PSC_RST_ERR_STAT); in psc_dma_status_irq() [all …]
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/linux/include/sound/ |
H A D | soc_sdw_utils.h | 1 /* SPDX-License-Identifier: GPL-2.0-only */ 13 #include <sound/soc-acpi.h> 25 * - 0 - No speaker output 26 * - SOC_SDW_CODEC_SPKR - CODEC internal speaker 27 * - SOC_SDW_SIDECAR_AMPS - 2x Sidecar amplifiers + CODEC internal speaker 28 * - SOC_SDW_CODEC_SPKR | SOF_SIDECAR_AMPS - Not currently supported 33 #define SOC_SDW_UNUSED_DAI_ID -1 47 const bool direction[2]; /* playback & capture support */ 58 bool playback); 89 struct device *headset_codec_dev; /* only one headset per card */ [all …]
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/linux/sound/pci/ca0106/ |
H A D | ca0106.h | 1 /* 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 …]
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/linux/Documentation/devicetree/bindings/sound/ |
H A D | fsl,ssi.yaml | 1 # 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 …]
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/linux/sound/pci/emu10k1/ |
H A D | p16v.h | 1 /* 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 …]
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/linux/sound/isa/sb/ |
H A D | sb_mixer.c | 1 // SPDX-License-Identifier: GPL-2.0-or-later 23 dev_dbg(chip->card->dev, "mixer_write 0x%x 0x%x\n", reg, data); in snd_sbmixer_write() 36 dev_dbg(chip->card->dev, "mixer_read 0x%x 0x%x\n", reg, result); in snd_sbmixer_read() 47 int mask = (kcontrol->private_value >> 24) & 0xff; in snd_sbmixer_info_single() 49 uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; in snd_sbmixer_info_single() 50 uinfo->count = 1; in snd_sbmixer_info_single() 51 uinfo->value.integer.min = 0; in snd_sbmixer_info_single() 52 uinfo->value.integer.max = mask; in snd_sbmixer_info_single() 60 int reg = kcontrol->private_value & 0xff; in snd_sbmixer_get_single() 61 int shift = (kcontrol->private_value >> 16) & 0xff; in snd_sbmixer_get_single() [all …]
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/linux/sound/pci/ |
H A D | azt3328.h | 1 /* 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 …]
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/linux/sound/soc/sof/amd/ |
H A D | acp70.c | 1 // 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 …]
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H A D | acp63.c | 1 // 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 …]
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H A D | rembrandt.c | 1 // 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 …]
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/linux/sound/usb/line6/ |
H A D | pcm.h | 1 /* 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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