diff options
Diffstat (limited to 'Documentation/sound/soc')
| -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.rst | 3 | ||||
| -rw-r--r-- | Documentation/sound/soc/dpcm.rst | 32 | ||||
| -rw-r--r-- | Documentation/sound/soc/index.rst | 1 | ||||
| -rw-r--r-- | Documentation/sound/soc/machine.rst | 26 | ||||
| -rw-r--r-- | Documentation/sound/soc/usb.rst | 482 |
7 files changed, 541 insertions, 19 deletions
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 = &dsp_codec_params, .num_c2c_params = 1, @@ -80,7 +80,7 @@ file: .codec_name = "codec-3, .codec_dai_name = "codec-3-dai_name", .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 = &dsp_codec_params, .num_c2c_params = 1, diff --git a/Documentation/sound/soc/dapm.rst b/Documentation/sound/soc/dapm.rst index 14c4dc026e6b..73a42d5a9f30 100644 --- a/Documentation/sound/soc/dapm.rst +++ b/Documentation/sound/soc/dapm.rst @@ -35,6 +35,9 @@ The graph for the STM32MP1-DK1 sound card is shown in picture: :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 ================== 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/index.rst b/Documentation/sound/soc/index.rst index e57df2dab2fd..8bed8f8f48da 100644 --- a/Documentation/sound/soc/index.rst +++ b/Documentation/sound/soc/index.rst @@ -18,3 +18,4 @@ The documentation is spilt into the following sections:- jack dpcm codec-to-codec + usb 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/soc/usb.rst b/Documentation/sound/soc/usb.rst new file mode 100644 index 000000000000..94c12f9d9dd1 --- /dev/null +++ b/Documentation/sound/soc/usb.rst @@ -0,0 +1,482 @@ +================ +ASoC USB support +================ + +Overview +======== +In order to leverage the existing USB sound device support in ALSA, the +ASoC USB APIs are introduced to allow the subsystems to exchange +configuration information. + +One potential use case would be to support USB audio offloading, which is +an implementation that allows for an alternate power-optimized path in the audio +subsystem to handle the transfer of audio data over the USB bus. This would +let the main processor to stay in lower power modes for longer duration. The +following is an example design of how the ASoC and ALSA pieces can be connected +together to achieve this: + +:: + + USB | ASoC + | _________________________ + | | ASoC Platform card | + | |_________________________| + | | | + | ___V____ ____V____ + | |ASoC BE | |ASoC FE | + | |DAI LNK | |DAI LNK | + | |________| |_________| + | ^ ^ ^ + | | |________| + | ___V____ | + | |SoC-USB | | + ________ ________ | | | + |USB SND |<--->|USBSND |<------------>|________| | + |(card.c)| |offld |<---------- | + |________| |________|___ | | | + ^ ^ | | | ____________V_________ + | | | | | |IPC | + __ V_______________V_____ | | | |______________________| + |USB SND (endpoint.c) | | | | ^ + |_________________________| | | | | + ^ | | | ___________V___________ + | | | |->|audio DSP | + ___________V_____________ | | |_______________________| + |XHCI HCD |<- | + |_________________________| | + + +SoC USB driver +============== +Structures +---------- +``struct snd_soc_usb`` + + - ``list``: list head for SND SoC struct list + - ``component``: reference to ASoC component + - ``connection_status_cb``: callback to notify connection events + - ``update_offload_route_info``: callback to fetch selected USB sound card/PCM + device + - ``priv_data``: driver data + +The snd_soc_usb structure can be referenced using the ASoC platform card +device, or a USB device (udev->dev). This is created by the ASoC BE DAI +link, and the USB sound entity will be able to pass information to the +ASoC BE DAI link using this structure. + +``struct snd_soc_usb_device`` + + - ``card_idx``: sound card index associated with USB sound device + - ``chip_idx``: USB sound chip array index + - ``cpcm_idx``: capture pcm device indexes associated with the USB sound device + - ``ppcm_idx``: playback pcm device indexes associated with the USB sound device + - ``num_playback``: number of playback streams + - ``num_capture``: number of capture streams + - ``list``: list head for the USB sound device list + +The struct snd_soc_usb_device is created by the USB sound offload driver. +This will carry basic parameters/limitations that will be used to +determine the possible offloading paths for this USB audio device. + +Functions +--------- +.. code-block:: rst + + int snd_soc_usb_find_supported_format(int card_idx, + struct snd_pcm_hw_params *params, int direction) +.. + + - ``card_idx``: the index into the USB sound chip array. + - ``params``: Requested PCM parameters from the USB DPCM BE DAI link + - ``direction``: capture or playback + +**snd_soc_usb_find_supported_format()** ensures that the requested audio profile +being requested by the external DSP is supported by the USB device. + +Returns 0 on success, and -EOPNOTSUPP on failure. + +.. code-block:: rst + + int snd_soc_usb_connect(struct device *usbdev, struct snd_soc_usb_device *sdev) +.. + + - ``usbdev``: the usb device that was discovered + - ``sdev``: capabilities of the device + +**snd_soc_usb_connect()** notifies the ASoC USB DCPM BE DAI link of a USB +audio device detection. This can be utilized in the BE DAI +driver to keep track of available USB audio devices. This is intended +to be called by the USB offload driver residing in USB SND. + +Returns 0 on success, negative error code on failure. + +.. code-block:: rst + + int snd_soc_usb_disconnect(struct device *usbdev, struct snd_soc_usb_device *sdev) +.. + + - ``usbdev``: the usb device that was removed + - ``sdev``: capabilities to free + +**snd_soc_usb_disconnect()** notifies the ASoC USB DCPM BE DAI link of a USB +audio device removal. This is intended to be called by the USB offload +driver that resides in USB SND. + +.. code-block:: rst + + void *snd_soc_usb_find_priv_data(struct device *usbdev) +.. + + - ``usbdev``: the usb device to reference to find private data + +**snd_soc_usb_find_priv_data()** fetches the private data saved to the SoC USB +device. + +Returns pointer to priv_data on success, NULL on failure. + +.. code-block:: rst + + int snd_soc_usb_setup_offload_jack(struct snd_soc_component *component, + struct snd_soc_jack *jack) +.. + + - ``component``: ASoC component to add the jack + - ``jack``: jack component to populate + +**snd_soc_usb_setup_offload_jack()** is a helper to add a sound jack control to +the platform sound card. This will allow for consistent naming to be used on +designs that support USB audio offloading. Additionally, this will enable the +jack to notify of changes. + +Returns 0 on success, negative otherwise. + +.. code-block:: rst + + int snd_soc_usb_update_offload_route(struct device *dev, int card, int pcm, + int direction, enum snd_soc_usb_kctl path, + long *route) +.. + + - ``dev``: USB device to look up offload path mapping + - ``card``: USB sound card index + - ``pcm``: USB sound PCM device index + - ``direction``: direction to fetch offload routing information + - ``path``: kcontrol selector - pcm device or card index + - ``route``: mapping of sound card and pcm indexes for the offload path. This is + an array of two integers that will carry the card and pcm device indexes + in that specific order. This can be used as the array for the kcontrol + output. + +**snd_soc_usb_update_offload_route()** calls a registered callback to the USB BE DAI +link to fetch the information about the mapped ASoC devices for executing USB audio +offload for the device. ``route`` may be a pointer to a kcontrol value output array, +which carries values when the kcontrol is read. + +Returns 0 on success, negative otherwise. + +.. code-block:: rst + + struct snd_soc_usb *snd_soc_usb_allocate_port(struct snd_soc_component *component, + void *data); +.. + + - ``component``: DPCM BE DAI link component + - ``data``: private data + +**snd_soc_usb_allocate_port()** allocates a SoC USB device and populates standard +parameters that is used for further operations. + +Returns a pointer to struct soc_usb on success, negative on error. + +.. code-block:: rst + + void snd_soc_usb_free_port(struct snd_soc_usb *usb); +.. + + - ``usb``: SoC USB device to free + +**snd_soc_usb_free_port()** frees a SoC USB device. + +.. code-block:: rst + + void snd_soc_usb_add_port(struct snd_soc_usb *usb); +.. + + - ``usb``: SoC USB device to add + +**snd_soc_usb_add_port()** add an allocated SoC USB device to the SOC USB framework. +Once added, this device can be referenced by further operations. + +.. code-block:: rst + + void snd_soc_usb_remove_port(struct snd_soc_usb *usb); +.. + + - ``usb``: SoC USB device to remove + +**snd_soc_usb_remove_port()** removes a SoC USB device from the SoC USB framework. +After removing a device, any SOC USB operations would not be able to reference the +device removed. + +How to Register to SoC USB +-------------------------- +The ASoC DPCM USB BE DAI link is the entity responsible for allocating and +registering the SoC USB device on the component bind. Likewise, it will +also be responsible for freeing the allocated resources. An example can +be shown below: + +.. code-block:: rst + + static int q6usb_component_probe(struct snd_soc_component *component) + { + ... + data->usb = snd_soc_usb_allocate_port(component, 1, &data->priv); + if (!data->usb) + return -ENOMEM; + + usb->connection_status_cb = q6usb_alsa_connection_cb; + + ret = snd_soc_usb_add_port(usb); + if (ret < 0) { + dev_err(component->dev, "failed to add usb port\n"); + goto free_usb; + } + ... + } + + static void q6usb_component_remove(struct snd_soc_component *component) + { + ... + snd_soc_usb_remove_port(data->usb); + snd_soc_usb_free_port(data->usb); + } + + static const struct snd_soc_component_driver q6usb_dai_component = { + .probe = q6usb_component_probe, + .remove = q6usb_component_remove, + .name = "q6usb-dai-component", + ... + }; +.. + +BE DAI links can pass along vendor specific information as part of the +call to allocate the SoC USB device. This will allow any BE DAI link +parameters or settings to be accessed by the USB offload driver that +resides in USB SND. + +USB Audio Device Connection Flow +-------------------------------- +USB devices can be hotplugged into the USB ports at any point in time. +The BE DAI link should be aware of the current state of the physical USB +port, i.e. if there are any USB devices with audio interface(s) connected. +connection_status_cb() can be used to notify the BE DAI link of any change. + +This is called whenever there is a USB SND interface bind or remove event, +using snd_soc_usb_connect() or snd_soc_usb_disconnect(): + +.. code-block:: rst + + static void qc_usb_audio_offload_probe(struct snd_usb_audio *chip) + { + ... + snd_soc_usb_connect(usb_get_usb_backend(udev), sdev); + ... + } + + static void qc_usb_audio_offload_disconnect(struct snd_usb_audio *chip) + { + ... + snd_soc_usb_disconnect(usb_get_usb_backend(chip->dev), dev->sdev); + ... + } +.. + +In order to account for conditions where driver or device existence is +not guaranteed, USB SND exposes snd_usb_rediscover_devices() to resend the +connect events for any identified USB audio interfaces. Consider the +the following situation: + + **usb_audio_probe()** + | --> USB audio streams allocated and saved to usb_chip[] + | --> Propagate connect event to USB offload driver in USB SND + | --> **snd_soc_usb_connect()** exits as USB BE DAI link is not ready + + BE DAI link component probe + | --> DAI link is probed and SoC USB port is allocated + | --> The USB audio device connect event is missed + +To ensure connection events are not missed, **snd_usb_rediscover_devices()** +is executed when the SoC USB device is registered. Now, when the BE DAI +link component probe occurs, the following highlights the sequence: + + BE DAI link component probe + | --> DAI link is probed and SoC USB port is allocated + | --> SoC USB device added, and **snd_usb_rediscover_devices()** runs + + **snd_usb_rediscover_devices()** + | --> Traverses through usb_chip[] and for non-NULL entries issue + | **connection_status_cb()** + +In the case where the USB offload driver is unbound, while USB SND is ready, +the **snd_usb_rediscover_devices()** is called during module init. This allows +for the offloading path to also be enabled with the following flow: + + **usb_audio_probe()** + | --> USB audio streams allocated and saved to usb_chip[] + | --> Propagate connect event to USB offload driver in USB SND + | --> USB offload driver **NOT** ready! + + BE DAI link component probe + | --> DAI link is probed and SoC USB port is allocated + | --> No USB connect event due to missing USB offload driver + + USB offload driver probe + | --> **qc_usb_audio_offload_init()** + | --> Calls **snd_usb_rediscover_devices()** to notify of devices + +USB Offload Related Kcontrols +============================= +Details +------- +A set of kcontrols can be utilized by applications to help select the proper sound +devices to enable USB audio offloading. SoC USB exposes the get_offload_dev() +callback that designs can use to ensure that the proper indices are returned to the +application. + +Implementation +-------------- + +**Example:** + + **Sound Cards**: + + :: + + 0 [SM8250MTPWCD938]: sm8250 - SM8250-MTP-WCD9380-WSA8810-VA-D + SM8250-MTP-WCD9380-WSA8810-VA-DMIC + 1 [Seri ]: USB-Audio - Plantronics Blackwire 3225 Seri + Plantronics Plantronics Blackwire + 3225 Seri at usb-xhci-hcd.1.auto-1.1, + full sp + 2 [C320M ]: USB-Audio - Plantronics C320-M + Plantronics Plantronics C320-M at usb-xhci-hcd.1.auto-1.2, full speed + + **PCM Devices**: + + :: + + card 0: SM8250MTPWCD938 [SM8250-MTP-WCD9380-WSA8810-VA-D], device 0: MultiMedia1 (*) [] + Subdevices: 1/1 + Subdevice #0: subdevice #0 + card 0: SM8250MTPWCD938 [SM8250-MTP-WCD9380-WSA8810-VA-D], device 1: MultiMedia2 (*) [] + Subdevices: 1/1 + Subdevice #0: subdevice #0 + card 1: Seri [Plantronics Blackwire 3225 Seri], device 0: USB Audio [USB Audio] + Subdevices: 1/1 + Subdevice #0: subdevice #0 + card 2: C320M [Plantronics C320-M], device 0: USB Audio [USB Audio] + Subdevices: 1/1 + Subdevice #0: subdevice #0 + + **USB Sound Card** - card#1: + + :: + + USB Offload Playback Card Route PCM#0 -1 (range -1->32) + USB Offload Playback PCM Route PCM#0 -1 (range -1->255) + + **USB Sound Card** - card#2: + + :: + + USB Offload Playback Card Route PCM#0 0 (range -1->32) + USB Offload Playback PCM Route PCM#0 1 (range -1->255) + +The above example shows a scenario where the system has one ASoC platform card +(card#0) and two USB sound devices connected (card#1 and card#2). When reading +the available kcontrols for each USB audio device, the following kcontrols lists +the mapped offload card and pcm device indexes for the specific USB device: + + ``USB Offload Playback Card Route PCM#*`` + + ``USB Offload Playback PCM Route PCM#*`` + +The kcontrol is indexed, because a USB audio device could potentially have +several PCM devices. The above kcontrols are defined as: + + - ``USB Offload Playback Card Route PCM#`` **(R)**: Returns the ASoC platform sound + card index for a mapped offload path. The output **"0"** (card index) signifies + that there is an available offload path for the USB SND device through card#0. + If **"-1"** is seen, then no offload path is available for the USB SND device. + This kcontrol exists for each USB audio device that exists in the system, and + its expected to derive the current status of offload based on the output value + for the kcontrol along with the PCM route kcontrol. + + - ``USB Offload Playback PCM Route PCM#`` **(R)**: Returns the ASoC platform sound + PCM device index for a mapped offload path. The output **"1"** (PCM device index) + signifies that there is an available offload path for the USB SND device through + PCM device#0. If **"-1"** is seen, then no offload path is available for the USB\ + SND device. This kcontrol exists for each USB audio device that exists in the + system, and its expected to derive the current status of offload based on the + output value for this kcontrol, in addition to the card route kcontrol. + +USB Offload Playback Route Kcontrol +----------------------------------- +In order to allow for vendor specific implementations on audio offloading device +selection, the SoC USB layer exposes the following: + +.. code-block:: rst + + int (*update_offload_route_info)(struct snd_soc_component *component, + int card, int pcm, int direction, + enum snd_soc_usb_kctl path, + long *route) +.. + +These are specific for the **USB Offload Playback Card Route PCM#** and **USB +Offload PCM Route PCM#** kcontrols. + +When users issue get calls to the kcontrol, the registered SoC USB callbacks will +execute the registered function calls to the DPCM BE DAI link. + +**Callback Registration:** + +.. code-block:: rst + + static int q6usb_component_probe(struct snd_soc_component *component) + { + ... + usb = snd_soc_usb_allocate_port(component, 1, &data->priv); + if (IS_ERR(usb)) + return -ENOMEM; + + usb->connection_status_cb = q6usb_alsa_connection_cb; + usb->update_offload_route_info = q6usb_get_offload_dev; + + ret = snd_soc_usb_add_port(usb); +.. + +Existing USB Sound Kcontrol +--------------------------- +With the introduction of USB offload support, the above USB offload kcontrol +will be added to the pre existing list of kcontrols identified by the USB sound +framework. These kcontrols are still the main controls that are used to +modify characteristics pertaining to the USB audio device. + + :: + + Number of controls: 9 + ctl type num name value + 0 INT 2 Capture Channel Map 0, 0 (range 0->36) + 1 INT 2 Playback Channel Map 0, 0 (range 0->36) + 2 BOOL 1 Headset Capture Switch On + 3 INT 1 Headset Capture Volume 10 (range 0->13) + 4 BOOL 1 Sidetone Playback Switch On + 5 INT 1 Sidetone Playback Volume 4096 (range 0->8192) + 6 BOOL 1 Headset Playback Switch On + 7 INT 2 Headset Playback Volume 20, 20 (range 0->24) + 8 INT 1 USB Offload Playback Card Route PCM#0 0 (range -1->32) + 9 INT 1 USB Offload Playback PCM Route PCM#0 1 (range -1->255) + +Since USB audio device controls are handled over the USB control endpoint, use the +existing mechanisms present in the USB mixer to set parameters, such as volume. |