diff options
Diffstat (limited to 'Documentation/sound')
| -rw-r--r-- | Documentation/sound/alsa-configuration.rst | 5 | ||||
| -rw-r--r-- | Documentation/sound/codecs/cs35l56.rst | 292 | ||||
| -rw-r--r-- | Documentation/sound/codecs/index.rst | 9 | ||||
| -rw-r--r-- | Documentation/sound/designs/compress-accel.rst | 134 | ||||
| -rw-r--r-- | Documentation/sound/designs/index.rst | 1 | ||||
| -rw-r--r-- | Documentation/sound/designs/midi-2.0.rst | 18 | ||||
| -rw-r--r-- | Documentation/sound/designs/powersave.rst | 6 | ||||
| -rw-r--r-- | Documentation/sound/hd-audio/notes.rst | 24 | ||||
| -rw-r--r-- | Documentation/sound/index.rst | 2 | ||||
| -rw-r--r-- | Documentation/sound/kernel-api/writing-an-alsa-driver.rst | 29 | ||||
| -rw-r--r-- | Documentation/sound/soc/clocking.rst | 12 | ||||
| -rw-r--r-- | Documentation/sound/soc/codec-to-codec.rst | 4 | ||||
| -rw-r--r-- | Documentation/sound/soc/dapm-graph.svg | 375 | ||||
| -rw-r--r-- | Documentation/sound/soc/dapm.rst | 170 | ||||
| -rw-r--r-- | Documentation/sound/soc/dpcm.rst | 32 | ||||
| -rw-r--r-- | Documentation/sound/soc/machine.rst | 26 | ||||
| -rw-r--r-- | Documentation/sound/utimers.rst | 126 |
17 files changed, 1146 insertions, 119 deletions
diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst index 829c672d9fe6..a45174d165eb 100644 --- a/Documentation/sound/alsa-configuration.rst +++ b/Documentation/sound/alsa-configuration.rst @@ -58,7 +58,7 @@ debug 2 = verbose debug messages); This option appears only when ``CONFIG_SND_DEBUG=y``. This option can be dynamically changed via sysfs - /sys/modules/snd/parameters/debug file. + /sys/module/snd/parameters/debug file. Module snd-pcm-oss ------------------ @@ -1059,6 +1059,9 @@ power_save Automatic power-saving timeout (in second, 0 = disable) power_save_controller Reset HD-audio controller in power-saving mode (default = on) +pm_blacklist + Enable / disable power-management deny-list (default = look up PM + deny-list, 0 = skip PM deny-list, 1 = force to turn off runtime PM) align_buffer_size Force rounding of buffer/period sizes to multiples of 128 bytes. This is more efficient in terms of memory access but isn't diff --git a/Documentation/sound/codecs/cs35l56.rst b/Documentation/sound/codecs/cs35l56.rst new file mode 100644 index 000000000000..98c6f6c74394 --- /dev/null +++ b/Documentation/sound/codecs/cs35l56.rst @@ -0,0 +1,292 @@ +.. SPDX-License-Identifier: GPL-2.0-only + +===================================================================== +Audio drivers for Cirrus Logic CS35L54/56/57 Boosted Smart Amplifiers +===================================================================== +:Copyright: 2025 Cirrus Logic, Inc. and + Cirrus Logic International Semiconductor Ltd. + +Contact: patches@opensource.cirrus.com + +Summary +======= + +The high-level summary of this document is: + +**If you have a laptop that uses CS35L54/56/57 amplifiers but audio is not +working, DO NOT ATTEMPT TO USE FIRMWARE AND SETTINGS FROM ANOTHER LAPTOP, +EVEN IF THAT LAPTOP SEEMS SIMILAR.** + +The CS35L54/56/57 amplifiers must be correctly configured for the power +supply voltage, speaker impedance, maximum speaker voltage/current, and +other external hardware connections. + +The amplifiers feature advanced boost technology that increases the voltage +used to drive the speakers, while proprietary speaker protection algorithms +allow these boosted amplifiers to push the limits of the speakers without +causing damage. These **must** be configured correctly. + +Supported Cirrus Logic amplifiers +--------------------------------- + +The cs35l56 drivers support: + +* CS35L54 +* CS35L56 +* CS35L57 + +There are two drivers in the kernel + +*For systems using SoundWire*: sound/soc/codecs/cs35l56.c and associated files + +*For systems using HDA*: sound/pci/hda/cs35l56_hda.c + +Firmware +======== + +The amplifier is controlled and managed by firmware running on the internal +DSP. Firmware files are essential to enable the full capabilities of the +amplifier. + +Firmware is distributed in the linux-firmware repository: +https://gitlab.com/kernel-firmware/linux-firmware.git + +On most SoundWire systems the amplifier has a default minimum capability to +produce audio. However this will be + +* at low volume, to protect the speakers, since the speaker specifications + and power supply voltages are unknown. +* a mono mix of left and right channels. + +On some SoundWire systems that have both CS42L43 and CS35L56/57 the CS35L56/57 +receive their audio from the CS42L43 instead of directly from the host +SoundWire interface. These systems can be identified by the CS42L43 showing +in dmesg as a SoundWire device, but the CS35L56/57 as SPI. On these systems +the firmware is *mandatory* to enable receiving the audio from the CS42L43. + +On HDA systems the firmware is *mandatory* to enable HDA bridge mode. There +will not be any audio from the amplifiers without firmware. + +Cirrus Logic firmware files +--------------------------- + +Each amplifier requires two firmware files. One file has a .wmfw suffix, the +other has a .bin suffix. + +The firmware is customized by the OEM to match the hardware of each laptop, +and the firmware is specific to that laptop. Because of this, there are many +firmware files in linux-firmware for these amplifiers. Firmware files are +**not interchangeable between laptops**. + +Cirrus Logic submits files for known laptops to the upstream linux-firmware +repository. Providing Cirrus Logic is aware of a particular laptop and has +permission from the manufacturer to publish the firmware, it will be pushed +to linux-firmware. You may need to upgrade to a newer release of +linux-firmware to obtain the firmware for your laptop. + +**Important:** the Makefile for linux-firmware creates symlinks that are listed +in the WHENCE file. These symlinks are required for the CS35L56 driver to be +able to load the firmware. + +How do I know which firmware file I should have? +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +All firmware file names are qualified with a unique "system ID". On normal +x86 PCs with PCI audio this is the Vendor Subsystem ID (SSID) of the host +PCI audio interface. + +The SSID can be viewed using the lspci tool:: + + lspci -v -nn | grep -A2 -i audio + 0000:00:1f.3 Audio device [0403]: Intel Corporation Meteor Lake-P HD Audio Controller [8086:7e28] + Subsystem: Dell Meteor Lake-P HD Audio Controller [1028:0c63] + +In this example the SSID is 10280c63. + +The format of the firmware file names is: + + cs35lxx-b0-dsp1-misc-SSID[-spkidX]-ampN + +Where: + + * cs35lxx-b0 is the amplifier model and silicon revision. This information + is logged by the driver during initialization. + * SSID is the 8-digit hexadecimal SSID value. + * ampN is the amplifier number (for example amp1). This is the same as + the prefix on the ALSA control names except that it is always lower-case + in the file name. + * spkidX is an optional part, used for laptops that have firmware + configurations for different makes and models of internal speakers. + +Sound Open Firmware and ALSA topology files +------------------------------------------- + +All SoundWire systems will require a Sound Open Firmware (SOF) for the +host CPU audio DSP, together with an ALSA topology file (.tplg). + +The SOF firmware will usually be provided by the manufacturer of the host +CPU (i.e. Intel or AMD). The .tplg file is normally part of the SOF firmware +release. + +SOF binary builds are available from: https://github.com/thesofproject/sof-bin/releases + +The main SOF source is here: https://github.com/thesofproject + +ALSA-ucm configurations +----------------------- +Typically an appropriate ALSA-ucm configuration file is needed for +use-case managers and audio servers such as PipeWire. + +Configuration files are available from the alsa-ucm-conf repository: +https://git.alsa-project.org/?p=alsa-ucm-conf.git + +Kernel log messages +=================== + +SoundWire +--------- +A successful initialization will look like this (this will be repeated for +each amplifier):: + + [ 7.568374] cs35l56 sdw:0:0:01fa:3556:01:0: supply VDD_P not found, using dummy regulator + [ 7.605208] cs35l56 sdw:0:0:01fa:3556:01:0: supply VDD_IO not found, using dummy regulator + [ 7.605313] cs35l56 sdw:0:0:01fa:3556:01:0: supply VDD_A not found, using dummy regulator + [ 7.939279] cs35l56 sdw:0:0:01fa:3556:01:0: Cirrus Logic CS35L56 Rev B0 OTP3 fw:3.4.4 (patched=0) + [ 7.947844] cs35l56 sdw:0:0:01fa:3556:01:0: Slave 4 state check1: UNATTACHED, status was 1 + [ 8.740280] cs35l56 sdw:0:0:01fa:3556:01:0: supply VDD_B not found, using dummy regulator + [ 8.740552] cs35l56 sdw:0:0:01fa:3556:01:0: supply VDD_AMP not found, using dummy regulator + [ 9.242164] cs35l56 sdw:0:0:01fa:3556:01:0: DSP1: cirrus/cs35l56-b0-dsp1-misc-xxxxxxxx.wmfw: format 3 timestamp 0x66b2b872 + [ 9.242173] cs35l56 sdw:0:0:01fa:3556:01:0: DSP1: cirrus/cs35l56-b0-dsp1-misc-xxxxxxxx.wmfw: Tue 05 Dec 2023 21:37:21 GMT Standard Time + [ 9.991709] cs35l56 sdw:0:0:01fa:3556:01:0: DSP1: Firmware: 1a00d6 vendor: 0x2 v3.11.23, 41 algorithms + [10.039098] cs35l56 sdw:0:0:01fa:3556:01:0: DSP1: cirrus/cs35l56-b0-dsp1-misc-xxxxxxxx-amp1.bin: v3.11.23 + [10.879235] cs35l56 sdw:0:0:01fa:3556:01:0: Slave 4 state check1: UNATTACHED, status was 1 + [11.401536] cs35l56 sdw:0:0:01fa:3556:01:0: Calibration applied + +HDA +--- +A successful initialization will look like this (this will be repeated for +each amplifier):: + + [ 6.306475] cs35l56-hda i2c-CSC3556:00-cs35l56-hda.0: Cirrus Logic CS35L56 Rev B0 OTP3 fw:3.4.4 (patched=0) + [ 6.613892] cs35l56-hda i2c-CSC3556:00-cs35l56-hda.0: DSP system name: 'xxxxxxxx', amp name: 'AMP1' + [ 8.266660] snd_hda_codec_cs8409 ehdaudio0D0: bound i2c-CSC3556:00-cs35l56-hda.0 (ops cs35l56_hda_comp_ops [snd_hda_scodec_cs35l56]) + [ 8.287525] cs35l56-hda i2c-CSC3556:00-cs35l56-hda.0: DSP1: cirrus/cs35l56-b0-dsp1-misc-xxxxxxxx.wmfw: format 3 timestamp 0x66b2b872 + [ 8.287528] cs35l56-hda i2c-CSC3556:00-cs35l56-hda.0: DSP1: cirrus/cs35l56-b0-dsp1-misc-xxxxxxxx.wmfw: Tue 05 Dec 2023 21:37:21 GMT Standard Time + [ 9.984335] cs35l56-hda i2c-CSC3556:00-cs35l56-hda.0: DSP1: Firmware: 1a00d6 vendor: 0x2 v3.11.23, 41 algorithms + [10.085797] cs35l56-hda i2c-CSC3556:00-cs35l56-hda.0: DSP1: cirrus/cs35l56-b0-dsp1-misc-xxxxxxxx-amp1.bin: v3.11.23 + [10.655237] cs35l56-hda i2c-CSC3556:00-cs35l56-hda.0: Calibration applied + +Important messages +~~~~~~~~~~~~~~~~~~ +Cirrus Logic CS35L56 Rev B0 OTP3 fw:3.4.4 (patched=0) + Shows that the driver has been able to read device ID registers from the + amplifier. + + * The actual amplifier type and silicon revision (CS35L56 B0 in this + example) is shown, as read from the amplifier identification registers. + * (patched=0) is normal, and indicates that the amplifier has been hard + reset and is running default ROM firmware. + * (patched=1) means that something has previously downloaded firmware + to the amplifier and the driver does not have control of the RESET + signal to be able to replace this preloaded firmware. This is normal + for systems where the BIOS downloads firmware to the amplifiers + before OS boot. + This status can also be seen if the cs35l56 kernel module is unloaded + and reloaded on a system where the driver does not have control of + RESET. SoundWire systems typically do not give the driver control of + RESET and only a BIOS (re)boot can reset the amplifiers. + +DSP1: cirrus/cs35l56-b0-dsp1-misc-xxxxxxxx.wmfw + Shows that a .wmfw firmware file was found and downloaded. + +DSP1: cirrus/cs35l56-b0-dsp1-misc-xxxxxxxx-amp1.bin + Shows that a .bin firmware file was found and downloaded. + +Calibration applied + Factory calibration data in EFI was written to the amplifier. + +Error messages +============== +This section explains some of the error messages that the driver can log. + +Algorithm coefficient version %d.%d.%d but expected %d.%d.%d + The version of the .bin file content does not match the loaded firmware. + Caused by mismatched .wmfw and .bin file, or .bin file was found but + .wmfw was not. + +No %s for algorithm %x + The version of the .bin file content does not match the loaded firmware. + Caused by mismatched .wmfw and .bin file, or .bin file was found but + .wmfw was not. + +.bin file required but not found + HDA driver did not find a .bin file that matches this hardware. + +Calibration disabled due to missing firmware controls + Driver was not able to write EFI calibration data to firmware registers. + This typically means that either: + + * The driver did not find a suitable wmfw for this hardware, or + * The amplifier has already been patched with firmware by something + previously, and the driver does not have control of a hard RESET line + to be able to reset the amplifier and download the firmware files it + found. This situation is indicated by the device identification + string in the kernel log shows "(patched=1)" + +Failed to write calibration + Same meaning and cause as "Calibration disabled due to missing firmware + controls" + +Failed to read calibration data from EFI + Factory calibration data in EFI is missing, empty or corrupt. + This is most likely to be cause by accidentally deleting the file from + the EFI filesystem. + +No calibration for silicon ID + The factory calibration data in EFI does not match this hardware. + The most likely cause is that an amplifier has been replaced on the + motherboard without going through manufacturer calibration process to + generate calibration data for the new amplifier. + +Did not find any buses for CSCxxxx + Only on HDA systems. The HDA codec driver found an ACPI entry for + Cirrus Logic companion amps, but could not enumerate the ACPI entries for + the I2C/SPI buses. The most likely cause of this is that: + + * The relevant bus driver (I2C or SPI) is not part of the kernel. + * The HDA codec driver was built-in to the kernel but the I2C/SPI + bus driver is a module and so the HDA codec driver cannot call the + bus driver functions. + +init_completion timed out + The SoundWire bus controller (host end) did not enumerate the amplifier. + In other words, the ACPI says there is an amplifier but for some reason + it was not detected on the bus. + +No AF01 node + Indicates an error in ACPI. A SoundWire system should have a Device() + node named "AF01" but it was not found. + +Failed to get spk-id-gpios + ACPI says that the driver should request a GPIO but the driver was not + able to get that GPIO. The most likely cause is that the kernel does not + include the correct GPIO or PINCTRL driver for this system. + +Failed to read spk-id + ACPI says that the driver should request a GPIO but the driver was not + able to read that GPIO. + +Unexpected spk-id element count + AF01 contains more speaker ID GPIO entries than the driver supports + +Overtemp error + Amplifier overheat protection was triggered and the amplifier shut down + to protect itself. + +Amp short error + Amplifier detected a short-circuit on the speaker output pins and shut + down for protection. This would normally indicate a damaged speaker. + +Hibernate wake failed + The driver tried to wake the amplifier from its power-saving state but + did not see the expected responses from the amplifier. This can be caused + by using firmware that does not match the hardware. diff --git a/Documentation/sound/codecs/index.rst b/Documentation/sound/codecs/index.rst new file mode 100644 index 000000000000..2cb95d87bbef --- /dev/null +++ b/Documentation/sound/codecs/index.rst @@ -0,0 +1,9 @@ +.. SPDX-License-Identifier: GPL-2.0 + +Codec-Specific Information +========================== + +.. toctree:: + :maxdepth: 2 + + cs35l56 diff --git a/Documentation/sound/designs/compress-accel.rst b/Documentation/sound/designs/compress-accel.rst new file mode 100644 index 000000000000..c9c1744b94c2 --- /dev/null +++ b/Documentation/sound/designs/compress-accel.rst @@ -0,0 +1,134 @@ +================================== +ALSA Co-processor Acceleration API +================================== + +Jaroslav Kysela <perex@perex.cz> + + +Overview +======== + +There is a requirement to expose the audio hardware that accelerates various +tasks for user space such as sample rate converters, compressed +stream decoders, etc. + +This is description for the API extension for the compress ALSA API which +is able to handle "tasks" that are not bound to real-time operations +and allows for the serialization of operations. + +Requirements +============ + +The main requirements are: + +- serialization of multiple tasks for user space to allow multiple + operations without user space intervention + +- separate buffers (input + output) for each operation + +- expose buffers using mmap to user space + +- signal user space when the task is finished (standard poll mechanism) + +Design +====== + +A new direction SND_COMPRESS_ACCEL is introduced to identify +the passthrough API. + +The API extension shares device enumeration and parameters handling from +the main compressed API. All other realtime streaming ioctls are deactivated +and a new set of task related ioctls are introduced. The standard +read/write/mmap I/O operations are not supported in the passthrough device. + +Device ("stream") state handling is reduced to OPEN/SETUP. All other +states are not available for the passthrough mode. + +Data I/O mechanism is using standard dma-buf interface with all advantages +like mmap, standard I/O, buffer sharing etc. One buffer is used for the +input data and second (separate) buffer is used for the output data. Each task +have separate I/O buffers. + +For the buffering parameters, the fragments means a limit of allocated tasks +for given device. The fragment_size limits the input buffer size for the given +device. The output buffer size is determined by the driver (may be different +from the input buffer size). + +State Machine +============= + +The passthrough audio stream state machine is described below:: + + +----------+ + | | + | OPEN | + | | + +----------+ + | + | + | compr_set_params() + | + v + all passthrough task ops +----------+ + +------------------------------------| | + | | SETUP | + | | + | +----------+ + | | + +------------------------------------------+ + + +Passthrough operations (ioctls) +=============================== + +All operations are protected using stream->device->lock (mutex). + +CREATE +------ +Creates a set of input/output buffers. The input buffer size is +fragment_size. Allocates unique seqno. + +The hardware drivers allocate internal 'struct dma_buf' for both input and +output buffers (using 'dma_buf_export()' function). The anonymous +file descriptors for those buffers are passed to user space. + +FREE +---- +Free a set of input/output buffers. If a task is active, the stop +operation is executed before. If seqno is zero, operation is executed for all +tasks. + +START +----- +Starts (queues) a task. There are two cases of the task start - right after +the task is created. In this case, origin_seqno must be zero. +The second case is for reusing of already finished task. The origin_seqno +must identify the task to be reused. In both cases, a new seqno value +is allocated and returned to user space. + +The prerequisite is that application filled input dma buffer with +new source data and set input_size to pass the real data size to the driver. + +The order of data processing is preserved (first started job must be +finished at first). + +If the multiple tasks require a state handling (e.g. resampling operation), +the user space may set SND_COMPRESS_TFLG_NEW_STREAM flag to mark the +start of the new stream data. It is useful to keep the allocated buffers +for the new operation rather using open/close mechanism. + +STOP +---- +Stop (dequeues) a task. If seqno is zero, operation is executed for all +tasks. + +STATUS +------ +Obtain the task status (active, finished). Also, the driver will set +the real output data size (valid area in the output buffer). + +Credits +======= +- Shengjiu Wang <shengjiu.wang@gmail.com> +- Takashi Iwai <tiwai@suse.de> +- Vinod Koul <vkoul@kernel.org> diff --git a/Documentation/sound/designs/index.rst b/Documentation/sound/designs/index.rst index b79db9ad8732..6b825c5617fc 100644 --- a/Documentation/sound/designs/index.rst +++ b/Documentation/sound/designs/index.rst @@ -6,6 +6,7 @@ Designs and Implementations control-names channel-mapping-api + compress-accel compress-offload timestamping jack-controls diff --git a/Documentation/sound/designs/midi-2.0.rst b/Documentation/sound/designs/midi-2.0.rst index 086487ca7ab1..71a343c93fe7 100644 --- a/Documentation/sound/designs/midi-2.0.rst +++ b/Documentation/sound/designs/midi-2.0.rst @@ -293,6 +293,17 @@ Rawmidi API Extensions status 0x05). When UMP core receives such a message, it updates the UMP EP info and the corresponding sequencer clients as well. +* The legacy rawmidi device number is found in the new `tied_device` + field of the rawmidi info. + On the other hand, the UMP rawmidi device number is found in + `tied_device` field of the legacy rawmidi info, too. + +* Each substream of the legacy rawmidi may be enabled / disabled + dynamically depending on the UMP FB state. + When the selected substream is inactive, it's indicated by the bit + 0x10 (`SNDRV_RAWMIDI_INFO_STREAM_INACTIVE`) in the `flags` field of + the legacy rawmidi info. + Control API Extensions ====================== @@ -377,6 +388,13 @@ Sequencer API Extensions announcement to the ALSA sequencer system port, similarly like the normal port change notification. +* There are two extended event types for notifying the UMP Endpoint and + Function Block changes via the system announcement port: + type 68 (`SNDRV_SEQ_EVENT_UMP_EP_CHANGE`) and type 69 + (`SNDRV_SEQ_EVENT_UMP_BLOCK_CHANGE`). They take the new type, + `snd_seq_ev_ump_notify` in the payload, indicating the client number + and the FB number that are changed. + MIDI2 USB Gadget Function Driver ================================ diff --git a/Documentation/sound/designs/powersave.rst b/Documentation/sound/designs/powersave.rst index 138157452eb9..ca7d1e838b4d 100644 --- a/Documentation/sound/designs/powersave.rst +++ b/Documentation/sound/designs/powersave.rst @@ -25,15 +25,15 @@ operations. The ``power_save`` option is exported as writable. This means you can adjust the value via sysfs on the fly. For example, to turn on the automatic power-save mode with 10 seconds, write to -``/sys/modules/snd_ac97_codec/parameters/power_save`` (usually as root): +``/sys/module/snd_ac97_codec/parameters/power_save`` (usually as root): :: - # echo 10 > /sys/modules/snd_ac97_codec/parameters/power_save + # echo 10 > /sys/module/snd_ac97_codec/parameters/power_save Note that you might hear click noise/pop when changing the power state. Also, it often takes certain time to wake up from the -power-down to the active state. These are often hardly to fix, so +power-down to the active state. These are often hard to fix, so don't report extra bug reports unless you have a fix patch ;-) For HD-audio interface, there is another module option, diff --git a/Documentation/sound/hd-audio/notes.rst b/Documentation/sound/hd-audio/notes.rst index a9e35b1f87bd..f81e94d8f145 100644 --- a/Documentation/sound/hd-audio/notes.rst +++ b/Documentation/sound/hd-audio/notes.rst @@ -15,7 +15,7 @@ problem is broken BIOS, and the rest is the driver implementation. This document explains the brief trouble-shooting and debugging methods for the HD-audio hardware. -The HD-audio component consists of two parts: the controller chip and +The HD-audio component consists of two parts: the controller chip and the codec chips on the HD-audio bus. Linux provides a single driver for all controllers, snd-hda-intel. Although the driver name contains a word of a well-known hardware vendor, it's not specific to it but for @@ -42,7 +42,7 @@ If you are interested in the deep debugging of HD-audio, read the HD-audio specification at first. The specification is found on Intel's web page, for example: -* https://www.intel.com/standards/hdaudio/ +* https://www.intel.com/content/www/us/en/standards/high-definition-audio-specification.html HD-Audio Controller @@ -81,7 +81,7 @@ the wake-up timing. It wakes up a few samples before actually processing the data on the buffer. This caused a lot of problems, for example, with ALSA dmix or JACK. Since 2.6.27 kernel, the driver puts an artificial delay to the wake up timing. This delay is controlled -via ``bdl_pos_adj`` option. +via ``bdl_pos_adj`` option. When ``bdl_pos_adj`` is a negative value (as default), it's assigned to an appropriate value depending on the controller chip. For Intel @@ -144,7 +144,7 @@ see a regression wrt the sound quality (stuttering, etc) or a lock-up in the recent kernel, try to pass ``enable_msi=0`` option to disable MSI. If it works, you can add the known bad device to the blacklist defined in hda_intel.c. In such a case, please report and give the -patch back to the upstream developer. +patch back to the upstream developer. HD-Audio Codec @@ -321,12 +321,6 @@ Kernel Configuration -------------------- In general, I recommend you to enable the sound debug option, ``CONFIG_SND_DEBUG=y``, no matter whether you are debugging or not. -This enables snd_printd() macro and others, and you'll get additional -kernel messages at probing. - -In addition, you can enable ``CONFIG_SND_DEBUG_VERBOSE=y``. But this -will give you far more messages. Thus turn this on only when you are -sure to want it. Don't forget to turn on the appropriate ``CONFIG_SND_HDA_CODEC_*`` options. Note that each of them corresponds to the codec chip, not @@ -375,7 +369,7 @@ HD-Audio Reconfiguration ------------------------ This is an experimental feature to allow you re-configure the HD-audio codec dynamically without reloading the driver. The following sysfs -files are available under each codec-hwdep device directory (e.g. +files are available under each codec-hwdep device directory (e.g. /sys/class/sound/hwC0D0): vendor_id @@ -433,7 +427,7 @@ re-configure based on that state, run like below: :: # echo 0x14 0x9993013f > /sys/class/sound/hwC0D0/user_pin_configs - # echo 1 > /sys/class/sound/hwC0D0/reconfig + # echo 1 > /sys/class/sound/hwC0D0/reconfig Hint Strings @@ -494,7 +488,7 @@ indep_hp (bool) mixer control, if available add_stereo_mix_input (bool) add the stereo mix (analog-loopback mix) to the input mux if - available + available add_jack_modes (bool) add "xxx Jack Mode" enum controls to each I/O jack for allowing to change the headphone amp and mic bias VREF capabilities @@ -504,7 +498,7 @@ power_save_node (bool) stream states power_down_unused (bool) power down the unused widgets, a subset of power_save_node, and - will be dropped in future + will be dropped in future add_hp_mic (bool) add the headphone to capture source if possible hp_mic_detect (bool) @@ -603,7 +597,7 @@ present. The patch module option is specific to each card instance, and you need to give one file name for each instance, separated by commas. -For example, if you have two cards, one for an on-board analog and one +For example, if you have two cards, one for an on-board analog and one for an HDMI video board, you may pass patch option like below: :: diff --git a/Documentation/sound/index.rst b/Documentation/sound/index.rst index 7e67e12730d3..51cd736f65b5 100644 --- a/Documentation/sound/index.rst +++ b/Documentation/sound/index.rst @@ -13,6 +13,8 @@ Sound Subsystem Documentation alsa-configuration hd-audio/index cards/index + codecs/index + utimers .. only:: subproject and html diff --git a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst index 2d2998faff62..895752cbcedd 100644 --- a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst +++ b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst @@ -3976,7 +3976,7 @@ Driver with A Single Source File Suppose you have a file xyz.c. Add the following two lines:: - snd-xyz-objs := xyz.o + snd-xyz-y := xyz.o obj-$(CONFIG_SND_XYZ) += snd-xyz.o 2. Create the Kconfig entry @@ -4019,7 +4019,7 @@ located in the new subdirectory, sound/pci/xyz. 2. Under the directory ``sound/pci/xyz``, create a Makefile:: - snd-xyz-objs := xyz.o abc.o def.o + snd-xyz-y := xyz.o abc.o def.o obj-$(CONFIG_SND_XYZ) += snd-xyz.o 3. Create the Kconfig entry @@ -4030,31 +4030,6 @@ located in the new subdirectory, sound/pci/xyz. Useful Functions ================ -:c:func:`snd_printk()` and friends ----------------------------------- - -.. note:: This subsection describes a few helper functions for - decorating a bit more on the standard :c:func:`printk()` & co. - However, in general, the use of such helpers is no longer recommended. - If possible, try to stick with the standard functions like - :c:func:`dev_err()` or :c:func:`pr_err()`. - -ALSA provides a verbose version of the :c:func:`printk()` function. -If a kernel config ``CONFIG_SND_VERBOSE_PRINTK`` is set, this function -prints the given message together with the file name and the line of the -caller. The ``KERN_XXX`` prefix is processed as well as the original -:c:func:`printk()` does, so it's recommended to add this prefix, -e.g. snd_printk(KERN_ERR "Oh my, sorry, it's extremely bad!\\n"); - -There are also :c:func:`printk()`'s for debugging. -:c:func:`snd_printd()` can be used for general debugging purposes. -If ``CONFIG_SND_DEBUG`` is set, this function is compiled, and works -just like :c:func:`snd_printk()`. If the ALSA is compiled without -the debugging flag, it's ignored. - -:c:func:`snd_printdd()` is compiled in only when -``CONFIG_SND_DEBUG_VERBOSE`` is set. - :c:func:`snd_BUG()` ------------------- diff --git a/Documentation/sound/soc/clocking.rst b/Documentation/sound/soc/clocking.rst index 32122d6877a3..25d016ea8b65 100644 --- a/Documentation/sound/soc/clocking.rst +++ b/Documentation/sound/soc/clocking.rst @@ -42,5 +42,17 @@ rate, number of channels and word size) to save on power. It is also desirable to use the codec (if possible) to drive (or master) the audio clocks as it usually gives more accurate sample rates than the CPU. +ASoC provided clock APIs +------------------------ +.. kernel-doc:: sound/soc/soc-dai.c + :identifiers: snd_soc_dai_set_sysclk +.. kernel-doc:: sound/soc/soc-dai.c + :identifiers: snd_soc_dai_set_clkdiv + +.. kernel-doc:: sound/soc/soc-dai.c + :identifiers: snd_soc_dai_set_pll + +.. kernel-doc:: sound/soc/soc-dai.c + :identifiers: snd_soc_dai_set_bclk_ratio diff --git a/Documentation/sound/soc/codec-to-codec.rst b/Documentation/sound/soc/codec-to-codec.rst index 0418521b6e03..973c147d9d82 100644 --- a/Documentation/sound/soc/codec-to-codec.rst +++ b/Documentation/sound/soc/codec-to-codec.rst @@ -68,7 +68,7 @@ file: .codec_dai_name = "codec-2-dai_name", .platform_name = "samsung-i2s.0", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .ignore_suspend = 1, .c2c_params = 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Left->cs42l51.0-004a_Left PGA</title> +<path fill="none" stroke="black" d="M804.92,-470.99C804.27,-460.3 803.44,-446.5 802.71,-434.37"/> +<polygon fill="black" stroke="black" points="806.2,-434.03 802.1,-424.26 799.21,-434.45 806.2,-434.03"/> +</g> +<!-- cs42l51.0-004a_Right PGA --> +<g id="node27" class="node"> +<title>cs42l51.0-004a_Right PGA</title> +<polygon fill="#f2f2f2" stroke="#4d4d4d" points="688.5,-424 605.5,-424 605.5,-386 688.5,-386 688.5,-424"/> +<text text-anchor="middle" x="647" y="-408.8" font-family="sans-serif" font-size="14.00">Right PGA</text> +<text text-anchor="middle" x="647" y="-393.8" font-family="sans-serif" font-size="14.00">[pga]</text> +</g> +<!-- cs42l51.0-004a_PGA-ADC Mux Right->cs42l51.0-004a_Right PGA --> +<g id="edge19" class="edge"> +<title>cs42l51.0-004a_PGA-ADC Mux Right->cs42l51.0-004a_Right PGA</title> +<path fill="none" stroke="black" d="M643.87,-470.99C644.38,-460.3 645.05,-446.5 645.63,-434.37"/> +<polygon fill="black" stroke="black" 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text-anchor="middle" x="154" y="-567.8" font-family="sans-serif" font-size="14.00">RX</text> +<text text-anchor="middle" x="154" y="-552.8" font-family="sans-serif" font-size="14.00">[output]</text> +</g> +<!-- hdmi-audio-codec.1.auto_RX->hdmi-audio-codec.1.auto_Capture --> +<g id="edge20" class="edge"> +<title>hdmi-audio-codec.1.auto_RX->hdmi-audio-codec.1.auto_Capture</title> +<path fill="none" stroke="black" d="M154.25,-544.83C154.36,-537.13 154.49,-527.97 154.61,-519.42"/> +<polygon fill="black" stroke="black" points="158.1,-519.46 154.74,-509.41 151.11,-519.36 158.1,-519.46"/> +</g> +</g> +</svg> diff --git a/Documentation/sound/soc/dapm.rst b/Documentation/sound/soc/dapm.rst index c3154ce6e1b2..73a42d5a9f30 100644 --- a/Documentation/sound/soc/dapm.rst +++ b/Documentation/sound/soc/dapm.rst @@ -7,8 +7,8 @@ Description Dynamic Audio Power Management (DAPM) is designed to allow portable Linux devices to use the minimum amount of power within the audio -subsystem at all times. It is independent of other kernel PM and as -such, can easily co-exist with the other PM systems. +subsystem at all times. It is independent of other kernel power +management frameworks and, as such, can easily co-exist with them. DAPM is also completely transparent to all user space applications as all power switching is done within the ASoC core. No code changes or @@ -16,11 +16,32 @@ recompiling are required for user space applications. DAPM makes power switching decisions based upon any audio stream (capture/playback) activity and audio mixer settings within the device. -DAPM spans the whole machine. It covers power control within the entire -audio subsystem, this includes internal codec power blocks and machine -level power systems. +DAPM is based on two basic elements, called widgets and routes: -There are 4 power domains within DAPM + * a **widget** is every part of the audio hardware that can be enabled by + software when in use and disabled to save power when not in use + * a **route** is an interconnection between widgets that exists when sound + can flow from one widget to the other + +All DAPM power switching decisions are made automatically by consulting an +audio routing graph. This graph is specific to each sound card and spans +the whole sound card, so some DAPM routes connect two widgets belonging to +different components (e.g. the LINE OUT pin of a CODEC and the input pin of +an amplifier). + +The graph for the STM32MP1-DK1 sound card is shown in picture: + +.. kernel-figure:: dapm-graph.svg + :alt: Example DAPM graph + :align: center + +You can also generate compatible graph for your sound card using +`tools/sound/dapm-graph` utility. + +DAPM power domains +================== + +There are 4 power domains within DAPM: Codec bias domain VREF, VMID (core codec and audio power) @@ -47,17 +68,11 @@ Stream domain Enabled and disabled when stream playback/capture is started and stopped respectively. e.g. aplay, arecord. -All DAPM power switching decisions are made automatically by consulting an audio -routing map of the whole machine. This map is specific to each machine and -consists of the interconnections between every audio component (including -internal codec components). All audio components that effect power are called -widgets hereafter. - DAPM Widgets ============ -Audio DAPM widgets fall into a number of types:- +Audio DAPM widgets fall into a number of types: Mixer Mixes several analog signals into a single analog signal. @@ -141,14 +156,14 @@ Stream Widgets relate to the stream power domain and only consist of ADCs (analog to digital converters), DACs (digital to analog converters), AIF IN and AIF OUT. -Stream widgets have the following format:- +Stream widgets have the following format: :: SND_SOC_DAPM_DAC(name, stream name, reg, shift, invert), SND_SOC_DAPM_AIF_IN(name, stream, slot, reg, shift, invert) NOTE: the stream name must match the corresponding stream name in your codec -snd_soc_codec_dai. +snd_soc_dai_driver. e.g. stream widgets for HiFi playback and capture :: @@ -167,7 +182,7 @@ Path Domain Widgets ------------------- Path domain widgets have a ability to control or affect the audio signal or -audio paths within the audio subsystem. They have the following form:- +audio paths within the audio subsystem. They have the following form: :: SND_SOC_DAPM_PGA(name, reg, shift, invert, controls, num_controls) @@ -207,7 +222,7 @@ powered. e.g. A machine widget can have an optional call back. e.g. Jack connector widget for an external Mic that enables Mic Bias -when the Mic is inserted:-:: +when the Mic is inserted:: static int spitz_mic_bias(struct snd_soc_dapm_widget* w, int event) { @@ -221,7 +236,7 @@ when the Mic is inserted:-:: Codec (BIAS) Domain ------------------- -The codec bias power domain has no widgets and is handled by the codecs DAPM +The codec bias power domain has no widgets and is handled by the codec DAPM event handler. This handler is called when the codec powerstate is changed wrt to any stream event or by kernel PM events. @@ -229,17 +244,58 @@ to any stream event or by kernel PM events. Virtual Widgets --------------- -Sometimes widgets exist in the codec or machine audio map that don't have any +Sometimes widgets exist in the codec or machine audio graph that don't have any corresponding soft power control. In this case it is necessary to create a virtual widget - a widget with no control bits e.g. :: SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), -This can be used to merge to signal paths together in software. +This can be used to merge two signal paths together in software. + +Registering DAPM controls +========================= + +In many cases the DAPM widgets are implemented statically in a ``static +const struct snd_soc_dapm_widget`` array in a codec driver, and simply +declared via the ``dapm_widgets`` and ``num_dapm_widgets`` fields of the +``struct snd_soc_component_driver``. + +Similarly, routes connecting them are implemented statically in a ``static +const struct snd_soc_dapm_route`` array and declared via the +``dapm_routes`` and ``num_dapm_routes`` fields of the same struct. + +With the above declared, the driver registration will take care of +populating them:: + + static const struct snd_soc_dapm_widget wm2000_dapm_widgets[] = { + SND_SOC_DAPM_OUTPUT("SPKN"), + SND_SOC_DAPM_OUTPUT("SPKP"), + ... + }; + + /* Target, Path, Source */ + static const struct snd_soc_dapm_route wm2000_audio_map[] = { + { "SPKN", NULL, "ANC Engine" }, + { "SPKP", NULL, "ANC Engine" }, + ... + }; -After all the widgets have been defined, they can then be added to the DAPM -subsystem individually with a call to snd_soc_dapm_new_control(). + static const struct snd_soc_component_driver soc_component_dev_wm2000 = { + ... + .dapm_widgets = wm2000_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm2000_dapm_widgets), + .dapm_routes = wm2000_audio_map, + .num_dapm_routes = ARRAY_SIZE(wm2000_audio_map), + ... + }; + +In more complex cases the list of DAPM widgets and/or routes can be only +known at probe time. This happens for example when a driver supports +different models having a different set of features. In those cases +separate widgets and routes arrays implementing the case-specific features +can be registered programmatically by calling snd_soc_dapm_new_controls() +and snd_soc_dapm_add_routes(). Codec/DSP Widget Interconnections @@ -247,31 +303,29 @@ Codec/DSP Widget Interconnections Widgets are connected to each other within the codec, platform and machine by audio paths (called interconnections). Each interconnection must be defined in -order to create a map of all audio paths between widgets. +order to create a graph of all audio paths between widgets. This is easiest with a diagram of the codec or DSP (and schematic of the machine audio system), as it requires joining widgets together via their audio signal paths. -e.g., from the WM8731 output mixer (wm8731.c) - -The WM8731 output mixer has 3 inputs (sources) +For example the WM8731 output mixer (wm8731.c) has 3 inputs (sources): 1. Line Bypass Input 2. DAC (HiFi playback) 3. Mic Sidetone Input -Each input in this example has a kcontrol associated with it (defined in example -above) and is connected to the output mixer via its kcontrol name. We can now -connect the destination widget (wrt audio signal) with its source widgets. -:: +Each input in this example has a kcontrol associated with it (defined in +the example above) and is connected to the output mixer via its kcontrol +name. We can now connect the destination widget (wrt audio signal) with its +source widgets. :: /* output mixer */ {"Output Mixer", "Line Bypass Switch", "Line Input"}, {"Output Mixer", "HiFi Playback Switch", "DAC"}, {"Output Mixer", "Mic Sidetone Switch", "Mic Bias"}, -So we have :- +So we have: * Destination Widget <=== Path Name <=== Source Widget, or * Sink, Path, Source, or @@ -280,12 +334,11 @@ So we have :- When there is no path name connecting widgets (e.g. a direct connection) we pass NULL for the path name. -Interconnections are created with a call to:- -:: +Interconnections are created with a call to:: snd_soc_dapm_connect_input(codec, sink, path, source); -Finally, snd_soc_dapm_new_widgets(codec) must be called after all widgets and +Finally, snd_soc_dapm_new_widgets() must be called after all widgets and interconnections have been registered with the core. This causes the core to scan the codec and machine so that the internal DAPM state matches the physical state of the machine. @@ -326,35 +379,44 @@ jacks can also be switched OFF. DAPM Widget Events ================== -Some widgets can register their interest with the DAPM core in PM events. -e.g. A Speaker with an amplifier registers a widget so the amplifier can be -powered only when the spk is in use. -:: +Widgets needing to implement a more complex behaviour than what DAPM can do +can set a custom "event handler" by setting a function pointer. An example +is a power supply needing to enable a GPIO:: - /* turn speaker amplifier on/off depending on use */ - static int corgi_amp_event(struct snd_soc_dapm_widget *w, int event) + static int sof_es8316_speaker_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) { - gpio_set_value(CORGI_GPIO_APM_ON, SND_SOC_DAPM_EVENT_ON(event)); - return 0; + if (SND_SOC_DAPM_EVENT_ON(event)) + gpiod_set_value_cansleep(gpio_pa, true); + else + gpiod_set_value_cansleep(gpio_pa, false); + + return 0; } - /* corgi machine dapm widgets */ - static const struct snd_soc_dapm_widget wm8731_dapm_widgets = - SND_SOC_DAPM_SPK("Ext Spk", corgi_amp_event); + static const struct snd_soc_dapm_widget st_widgets[] = { + ... + SND_SOC_DAPM_SUPPLY("Speaker Power", SND_SOC_NOPM, 0, 0, + sof_es8316_speaker_power_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + }; -Please see soc-dapm.h for all other widgets that support events. +See soc-dapm.h for all other widgets that support events. Event types ----------- -The following event types are supported by event widgets. -:: +The following event types are supported by event widgets:: /* dapm event types */ - #define SND_SOC_DAPM_PRE_PMU 0x1 /* before widget power up */ - #define SND_SOC_DAPM_POST_PMU 0x2 /* after widget power up */ - #define SND_SOC_DAPM_PRE_PMD 0x4 /* before widget power down */ - #define SND_SOC_DAPM_POST_PMD 0x8 /* after widget power down */ - #define SND_SOC_DAPM_PRE_REG 0x10 /* before audio path setup */ - #define SND_SOC_DAPM_POST_REG 0x20 /* after audio path setup */ + #define SND_SOC_DAPM_PRE_PMU 0x1 /* before widget power up */ + #define SND_SOC_DAPM_POST_PMU 0x2 /* after widget power up */ + #define SND_SOC_DAPM_PRE_PMD 0x4 /* before widget power down */ + #define SND_SOC_DAPM_POST_PMD 0x8 /* after widget power down */ + #define SND_SOC_DAPM_PRE_REG 0x10 /* before audio path setup */ + #define SND_SOC_DAPM_POST_REG 0x20 /* after audio path setup */ + #define SND_SOC_DAPM_WILL_PMU 0x40 /* called at start of sequence */ + #define SND_SOC_DAPM_WILL_PMD 0x80 /* called at start of sequence */ + #define SND_SOC_DAPM_PRE_POST_PMD (SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD) + #define SND_SOC_DAPM_PRE_POST_PMU (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU) diff --git a/Documentation/sound/soc/dpcm.rst b/Documentation/sound/soc/dpcm.rst index 2d7ad1d91504..7b6aeab3c207 100644 --- a/Documentation/sound/soc/dpcm.rst +++ b/Documentation/sound/soc/dpcm.rst @@ -147,25 +147,25 @@ For the example above we have to define 4 FE DAI links and 6 BE DAI links. The FE DAI links are defined as follows :- :: + SND_SOC_DAILINK_DEFS(pcm0, + DAILINK_COMP_ARRAY(COMP_CPU("System Pin")), + DAILINK_COMP_ARRAY(COMP_DUMMY()), + DAILINK_COMP_ARRAY(COMP_PLATFORM("dsp-audio"))); + static struct snd_soc_dai_link machine_dais[] = { { .name = "PCM0 System", .stream_name = "System Playback", - .cpu_dai_name = "System Pin", - .platform_name = "dsp-audio", - .codec_name = "snd-soc-dummy", - .codec_dai_name = "snd-soc-dummy-dai", + SND_SOC_DAILINK_REG(pcm0), .dynamic = 1, .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_playback = 1, }, .....< other FE and BE DAI links here > }; This FE DAI link is pretty similar to a regular DAI link except that we also -set the DAI link to a DPCM FE with the ``dynamic = 1``. The supported FE stream -directions should also be set with the ``dpcm_playback`` and ``dpcm_capture`` -flags. There is also an option to specify the ordering of the trigger call for +set the DAI link to a DPCM FE with the ``dynamic = 1``. +There is also an option to specify the ordering of the trigger call for each FE. This allows the ASoC core to trigger the DSP before or after the other components (as some DSPs have strong requirements for the ordering DAI/DSP start and stop sequences). @@ -176,28 +176,26 @@ dynamic and will change depending on runtime config. The BE DAIs are configured as follows :- :: + SND_SOC_DAILINK_DEFS(headset, + DAILINK_COMP_ARRAY(COMP_CPU("ssp-dai.0")), + DAILINK_COMP_ARRAY(COMP_CODEC("rt5640.0-001c", "rt5640-aif1"))); + static struct snd_soc_dai_link machine_dais[] = { .....< FE DAI links here > { .name = "Codec Headset", - .cpu_dai_name = "ssp-dai.0", - .platform_name = "snd-soc-dummy", + SND_SOC_DAILINK_REG(headset), .no_pcm = 1, - .codec_name = "rt5640.0-001c", - .codec_dai_name = "rt5640-aif1", .ignore_suspend = 1, .ignore_pmdown_time = 1, .be_hw_params_fixup = hswult_ssp0_fixup, .ops = &haswell_ops, - .dpcm_playback = 1, - .dpcm_capture = 1, }, .....< other BE DAI links here > }; This BE DAI link connects DAI0 to the codec (in this case RT5460 AIF1). It sets -the ``no_pcm`` flag to mark it has a BE and sets flags for supported stream -directions using ``dpcm_playback`` and ``dpcm_capture`` above. +the ``no_pcm`` flag to mark it has a BE. The BE has also flags set for ignoring suspend and PM down time. This allows the BE to work in a hostless mode where the host CPU is not transferring data @@ -367,7 +365,7 @@ The machine driver sets some additional parameters to the DAI link i.e. .codec_dai_name = "modem-aif1", .codec_name = "modem", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .c2c_params = &dai_params, .num_c2c_params = 1, } diff --git a/Documentation/sound/soc/machine.rst b/Documentation/sound/soc/machine.rst index 515c9444deaf..1828f5edca3e 100644 --- a/Documentation/sound/soc/machine.rst +++ b/Documentation/sound/soc/machine.rst @@ -71,6 +71,18 @@ struct snd_soc_dai_link is used to set up each DAI in your machine. e.g. .ops = &corgi_ops, }; +In the above struct, dai’s are registered using names but you can pass +either dai name or device tree node but not both. Also, names used here +for cpu/codec/platform dais should be globally unique. + +Additionally below example macro can be used to register cpu, codec and +platform dai:: + + SND_SOC_DAILINK_DEFS(wm2200_cpu_dsp, + DAILINK_COMP_ARRAY(COMP_CPU("samsung-i2s.0")), + DAILINK_COMP_ARRAY(COMP_CODEC("spi0.0", "wm0010-sdi1")), + DAILINK_COMP_ARRAY(COMP_PLATFORM("samsung-i2s.0"))); + struct snd_soc_card then sets up the machine with its DAIs. e.g. :: @@ -81,6 +93,10 @@ struct snd_soc_card then sets up the machine with its DAIs. e.g. .num_links = 1, }; +Following this, ``devm_snd_soc_register_card`` can be used to register +the sound card. During the registration, the individual components +such as the codec, CPU, and platform are probed. If all these components +are successfully probed, the sound card gets registered. Machine Power Map ----------------- @@ -95,3 +111,13 @@ Machine Controls ---------------- Machine specific audio mixer controls can be added in the DAI init function. + + +Clocking Controls +----------------- + +As previously noted, clock configuration is handled within the machine driver. +For details on the clock APIs that the machine driver can utilize for +setup, please refer to Documentation/sound/soc/clocking.rst. However, the +callback needs to be registered by the CPU/Codec/Platform drivers to configure +the clocks that is needed for the corresponding device operation. diff --git a/Documentation/sound/utimers.rst b/Documentation/sound/utimers.rst new file mode 100644 index 000000000000..ec21567d3f72 --- /dev/null +++ b/Documentation/sound/utimers.rst @@ -0,0 +1,126 @@ +.. SPDX-License-Identifier: GPL-2.0 + +======================= +Userspace-driven timers +======================= + +:Author: Ivan Orlov <ivan.orlov0322@gmail.com> + +Preface +======= + +This document describes the userspace-driven timers: virtual ALSA timers +which could be created and controlled by userspace applications using +IOCTL calls. Such timers could be useful when synchronizing audio +stream with timer sources which we don't have ALSA timers exported for +(e.g. PTP clocks), and when synchronizing the audio stream going through +two virtual sound devices using ``snd-aloop`` (for instance, when +we have a network application sending frames to one snd-aloop device, +and another sound application listening on the other end of snd-aloop). + +Enabling userspace-driven timers +================================ + +The userspace-driven timers could be enabled in the kernel using the +``CONFIG_SND_UTIMER`` configuration option. It depends on the +``CONFIG_SND_TIMER`` option, so it also should be enabled. + +Userspace-driven timers API +=========================== + +Userspace application can create a userspace-driven ALSA timer by +executing the ``SNDRV_TIMER_IOCTL_CREATE`` ioctl call on the +``/dev/snd/timer`` device file descriptor. The ``snd_timer_uinfo`` +structure should be passed as an ioctl argument: + +:: + + struct snd_timer_uinfo { + __u64 resolution; + int fd; + unsigned int id; + unsigned char reserved[16]; + } + +The ``resolution`` field sets the desired resolution in nanoseconds for +the virtual timer. ``resolution`` field simply provides an information +about the virtual timer, but does not affect the timing itself. ``id`` +field gets overwritten by the ioctl, and the identifier you get in this +field after the call can be used as a timer subdevice number when +passing the timer to ``snd-aloop`` kernel module or other userspace +applications. There could be up to 128 userspace-driven timers in the +system at one moment of time, thus the id value ranges from 0 to 127. + +Besides from overwriting the ``snd_timer_uinfo`` struct, ioctl stores +a timer file descriptor, which can be used to trigger the timer, in the +``fd`` field of the ``snd_timer_uinfo`` struct. Allocation of a file +descriptor for the timer guarantees that the timer can only be triggered +by the process which created it. The timer then can be triggered with +``SNDRV_TIMER_IOCTL_TRIGGER`` ioctl call on the timer file descriptor. + +So, the example code for creating and triggering the timer would be: + +:: + + static struct snd_timer_uinfo utimer_info = { + /* Timer is going to tick (presumably) every 1000000 ns */ + .resolution = 1000000ULL, + .id = -1, + }; + + int timer_device_fd = open("/dev/snd/timer", O_RDWR | O_CLOEXEC); + + if (ioctl(timer_device_fd, SNDRV_TIMER_IOCTL_CREATE, &utimer_info)) { + perror("Failed to create the timer"); + return -1; + } + + ... + + /* + * Now we want to trigger the timer. Callbacks of all of the + * timer instances binded to this timer will be executed after + * this call. + */ + ioctl(utimer_info.fd, SNDRV_TIMER_IOCTL_TRIGGER, NULL); + + ... + + /* Now, destroy the timer */ + close(timer_info.fd); + + +More detailed example of creating and ticking the timer could be found +in the utimer ALSA selftest. + +Userspace-driven timers and snd-aloop +------------------------------------- + +Userspace-driven timers could be easily used with ``snd-aloop`` module +when synchronizing two sound applications on both ends of the virtual +sound loopback. For instance, if one of the applications receives sound +frames from network and sends them to snd-aloop pcm device, and another +application listens for frames on the other snd-aloop pcm device, it +makes sense that the ALSA middle layer should initiate a data +transaction when the new period of data is received through network, but +not when the certain amount of jiffies elapses. Userspace-driven ALSA +timers could be used to achieve this. + +To use userspace-driven ALSA timer as a timer source of snd-aloop, pass +the following string as the snd-aloop ``timer_source`` parameter: + +:: + + # modprobe snd-aloop timer_source="-1.4.<utimer_id>" + +Where ``utimer_id`` is the id of the timer you created with +``SNDRV_TIMER_IOCTL_CREATE``, and ``4`` is the number of +userspace-driven timers device (``SNDRV_TIMER_GLOBAL_UDRIVEN``). + +``resolution`` for the userspace-driven ALSA timer used with snd-aloop +should be calculated as ``1000000000ULL / frame_rate * period_size`` as +the timer is going to tick every time a new period of frames is ready. + +After that, each time you trigger the timer with +``SNDRV_TIMER_IOCTL_TRIGGER`` the new period of data will be transferred +from one snd-aloop device to another. |
