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https://github.com/AuxXxilium/linux_dsm_epyc7002.git
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126f7051b4
We've got many code additions at this cycle as a result of quite a few new drivers. Below are highlights: Core stuff: - Fix the long-standing issue with the device registration order; the control device is now registered at last - PCM locking code cleanups for RT kernels - Fixes for possible races in ALSA timer resolution accesses - TLV offset definitions in uapi ASoC: - Many fixes for the topology stuff, including fixes for v4 ABI compatibility - Lots of cleanups / quirks for Intel platforms based on Realtek CODECs - Continued componentization works, removing legacy CODEC stuff - Conversion of OMAP DMA to the new, more standard SDMA-PCM driver - Fixes and updates to Cirrus Logic SoC drivers - New Qualcomm DSP support - New drivers for Analog SSM2305, Atmel I2S controllers, Mediatek MT6351, MT6797 and MT7622, Qualcomm DSPs, Realtek RT1305, RT1306 and RT5668 and TI TSCS454 HD-audio: - Finally better support for some CA0132 boards, allowing Windows firmware - HP Spectre x360 support along with a bulk of COEF stuff - Blacklisting power save default some known boards reported on Fedora USB-audio: - Continued improvements on UAC3 support; now BADD is supported - Fixes / improvements for Dell WD15 dock - Allow DMA coherent pages for PCM buffers for ARCH, MIPS & co Others: - New Xen sound frontend driver support - Cache implementation and other improvements for FireWire DICE - Conversions to octal permissions in allover places -----BEGIN PGP SIGNATURE----- iQJCBAABCAAsFiEEIXTw5fNLNI7mMiVaLtJE4w1nLE8FAlsXjEAOHHRpd2FpQHN1 c2UuZGUACgkQLtJE4w1nLE/szw/9FdtjD7LOBMNgbVbeU+SDTEUGh1OdIElSE+bL vU1USUNud9TgYb4SFM4grjsB9v6vM+bZ8mquzLpSzGj2J/Yl3dT7reyr6TYfoGfE oCETfYLk0gbhQrrqWoVwRHsPAJYyj6dGXZ/Kiy9MuD9bfWUGAehuqKl1inySxcGb VTrhlegHApRJ7z+Yzn6K3Git+aCYhpgxO5NK1DkoagHAsAhJhdavYWhm8lcQ4pnO UlahRms0cTpDFrIkHHKH+c1ihyxn3RVpueQIIpx5xRpIHXezMnUk8mpRduqcYGk2 9cxVlC4KMucsAud39joGN6BWlkmfpmlMfGkdVlAnlBpdTyYC1pJRCKPXX3P+d9Tk NvH3jKx/izNYFPLOysoV4vc5puDqSEfAC3geD+ugJFhhJuH9sAMyGOx9MRhC6ebf qGI2IEhyn9tVc8/f3glEqS79WDHas+dUCkXxhbAQtj4NcjqgXkM26h5MnrzIYV23 m5iAzXIDuT44Qw1BxHQb40DzgHZMU3p/c/PAqU/Ex9+Oi1mq6E8V7MH+n9Ylo2vN Mw3atYiLqv9xv+7/MmvGUQuTyMR3HB0KyNUCcSyuWPnuqZ/Gi+wIg9cuYXYfrn57 NtCoW4gzaex909z+QoZa5YxYRfZBJuRjYU0ugOBdK6R3+l/6IsGTasdR6LqngYY6 RIgK2S8= =37hC -----END PGP SIGNATURE----- Merge tag 'sound-4.18-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound updates from Takashi Iwai: "We've got many code additions at this cycle as a result of quite a few new drivers. Below are highlights: Core stuff: - Fix the long-standing issue with the device registration order; the control device is now registered at last - PCM locking code cleanups for RT kernels - Fixes for possible races in ALSA timer resolution accesses - TLV offset definitions in uapi ASoC: - Many fixes for the topology stuff, including fixes for v4 ABI compatibility - Lots of cleanups / quirks for Intel platforms based on Realtek CODECs - Continued componentization works, removing legacy CODEC stuff - Conversion of OMAP DMA to the new, more standard SDMA-PCM driver - Fixes and updates to Cirrus Logic SoC drivers - New Qualcomm DSP support - New drivers for Analog SSM2305, Atmel I2S controllers, Mediatek MT6351, MT6797 and MT7622, Qualcomm DSPs, Realtek RT1305, RT1306 and RT5668 and TI TSCS454 HD-audio: - Finally better support for some CA0132 boards, allowing Windows firmware - HP Spectre x360 support along with a bulk of COEF stuff - Blacklisting power save default some known boards reported on Fedora USB-audio: - Continued improvements on UAC3 support; now BADD is supported - Fixes / improvements for Dell WD15 dock - Allow DMA coherent pages for PCM buffers for ARCH, MIPS & co Others: - New Xen sound frontend driver support - Cache implementation and other improvements for FireWire DICE - Conversions to octal permissions in allover places" * tag 'sound-4.18-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (386 commits) ASoC: dapm: delete dapm_kcontrol_data paths list before freeing it ALSA: usb-audio: remove redundant check on err ASoC: topology: Move skl-tplg-interface.h to uapi ASoC: topology: Move v4 manifest header data structures to uapi ASoC: topology: Improve backwards compatibility with v4 topology files ALSA: pci/hda: Remove unused, broken, header file ASoC: TSCS454: Add Support ASoC: Intel: kbl: Move codec sysclk config to codec_init function ASoC: simple-card: set cpu dai clk in hw_params ALSA: hda - Handle kzalloc() failure in snd_hda_attach_pcm_stream() ALSA: oxygen: use match_string() helper ASoC: dapm: use match_string() helper ASoC: max98095: use match_string() helper ASoC: max98088: use match_string() helper ASoC: Intel: bytcr_rt5651: Set card long_name based on quirks ASoC: mt6797-mt6351: add hostless phone call path ASoC: mt6797: add Hostless DAI ASoC: mt6797: add PCM interface ASoC: mediatek: export mtk-afe symbols as needed ASoC: codecs: PCM1789: include gpio/consumer.h ...
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=======================
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ASoC Codec Class Driver
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=======================
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The codec class driver is generic and hardware independent code that configures
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the codec, FM, MODEM, BT or external DSP to provide audio capture and playback.
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It should contain no code that is specific to the target platform or machine.
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All platform and machine specific code should be added to the platform and
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machine drivers respectively.
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Each codec class driver *must* provide the following features:-
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1. Codec DAI and PCM configuration
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2. Codec control IO - using RegMap API
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3. Mixers and audio controls
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4. Codec audio operations
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5. DAPM description.
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6. DAPM event handler.
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Optionally, codec drivers can also provide:-
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7. DAC Digital mute control.
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Its probably best to use this guide in conjunction with the existing codec
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driver code in sound/soc/codecs/
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ASoC Codec driver breakdown
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===========================
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Codec DAI and PCM configuration
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-------------------------------
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Each codec driver must have a struct snd_soc_dai_driver to define its DAI and
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PCM capabilities and operations. This struct is exported so that it can be
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registered with the core by your machine driver.
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e.g.
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::
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static struct snd_soc_dai_ops wm8731_dai_ops = {
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.prepare = wm8731_pcm_prepare,
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.hw_params = wm8731_hw_params,
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.shutdown = wm8731_shutdown,
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.digital_mute = wm8731_mute,
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.set_sysclk = wm8731_set_dai_sysclk,
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.set_fmt = wm8731_set_dai_fmt,
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};
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struct snd_soc_dai_driver wm8731_dai = {
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.name = "wm8731-hifi",
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.playback = {
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.stream_name = "Playback",
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.channels_min = 1,
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.channels_max = 2,
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.rates = WM8731_RATES,
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.formats = WM8731_FORMATS,},
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.capture = {
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.stream_name = "Capture",
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.channels_min = 1,
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.channels_max = 2,
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.rates = WM8731_RATES,
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.formats = WM8731_FORMATS,},
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.ops = &wm8731_dai_ops,
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.symmetric_rates = 1,
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};
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Codec control IO
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----------------
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The codec can usually be controlled via an I2C or SPI style interface
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(AC97 combines control with data in the DAI). The codec driver should use the
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Regmap API for all codec IO. Please see include/linux/regmap.h and existing
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codec drivers for example regmap usage.
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Mixers and audio controls
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-------------------------
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All the codec mixers and audio controls can be defined using the convenience
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macros defined in soc.h.
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::
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#define SOC_SINGLE(xname, reg, shift, mask, invert)
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Defines a single control as follows:-
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::
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xname = Control name e.g. "Playback Volume"
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reg = codec register
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shift = control bit(s) offset in register
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mask = control bit size(s) e.g. mask of 7 = 3 bits
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invert = the control is inverted
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Other macros include:-
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::
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#define SOC_DOUBLE(xname, reg, shift_left, shift_right, mask, invert)
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A stereo control
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::
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#define SOC_DOUBLE_R(xname, reg_left, reg_right, shift, mask, invert)
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A stereo control spanning 2 registers
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::
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#define SOC_ENUM_SINGLE(xreg, xshift, xmask, xtexts)
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Defines an single enumerated control as follows:-
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::
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xreg = register
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xshift = control bit(s) offset in register
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xmask = control bit(s) size
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xtexts = pointer to array of strings that describe each setting
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#define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts)
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Defines a stereo enumerated control
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Codec Audio Operations
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----------------------
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The codec driver also supports the following ALSA PCM operations:-
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::
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/* SoC audio ops */
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struct snd_soc_ops {
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int (*startup)(struct snd_pcm_substream *);
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void (*shutdown)(struct snd_pcm_substream *);
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int (*hw_params)(struct snd_pcm_substream *, struct snd_pcm_hw_params *);
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int (*hw_free)(struct snd_pcm_substream *);
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int (*prepare)(struct snd_pcm_substream *);
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};
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Please refer to the ALSA driver PCM documentation for details.
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http://www.alsa-project.org/~iwai/writing-an-alsa-driver/
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DAPM description
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----------------
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The Dynamic Audio Power Management description describes the codec power
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components and their relationships and registers to the ASoC core.
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Please read dapm.rst for details of building the description.
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Please also see the examples in other codec drivers.
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DAPM event handler
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------------------
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This function is a callback that handles codec domain PM calls and system
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domain PM calls (e.g. suspend and resume). It is used to put the codec
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to sleep when not in use.
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Power states:-
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::
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SNDRV_CTL_POWER_D0: /* full On */
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/* vref/mid, clk and osc on, active */
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SNDRV_CTL_POWER_D1: /* partial On */
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SNDRV_CTL_POWER_D2: /* partial On */
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SNDRV_CTL_POWER_D3hot: /* Off, with power */
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/* everything off except vref/vmid, inactive */
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SNDRV_CTL_POWER_D3cold: /* Everything Off, without power */
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Codec DAC digital mute control
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------------------------------
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Most codecs have a digital mute before the DACs that can be used to
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minimise any system noise. The mute stops any digital data from
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entering the DAC.
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A callback can be created that is called by the core for each codec DAI
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when the mute is applied or freed.
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i.e.
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::
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static int wm8974_mute(struct snd_soc_dai *dai, int mute)
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{
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struct snd_soc_component *component = dai->component;
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u16 mute_reg = snd_soc_component_read32(component, WM8974_DAC) & 0xffbf;
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if (mute)
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snd_soc_component_write(component, WM8974_DAC, mute_reg | 0x40);
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else
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snd_soc_component_write(component, WM8974_DAC, mute_reg);
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return 0;
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}
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