diff options
61 files changed, 1751 insertions, 1899 deletions
diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst index 5f31fa5e2435..829c672d9fe6 100644 --- a/Documentation/sound/alsa-configuration.rst +++ b/Documentation/sound/alsa-configuration.rst @@ -133,6 +133,19 @@ enable enable card; Default: enabled, for PCI and ISA PnP cards +These options are used for either specifying the order of instances or +controlling enabling and disabling of each one of the devices if there +are multiple devices bound with the same driver. For example, there are +many machines which have two HD-audio controllers (one for HDMI/DP +audio and another for onboard analog). In most cases, the second one is +in primary usage, and people would like to assign it as the first +appearing card. They can do it by specifying "index=1,0" module +parameter, which will swap the assignment slots. + +Today, with the sound backend like PulseAudio and PipeWire which +supports dynamic configuration, it's of little use, but that was a +help for static configuration in the past. + Module snd-adlib ---------------- @@ -723,9 +736,10 @@ Module for EMU10K1/EMU10k2 based PCI sound cards. * Sound Blaster Live! * Sound Blaster PCI 512 -* Emu APS (partially supported) * Sound Blaster Audigy - +* E-MU APS (partially supported) +* E-MU DAS + extin bitmap of available external inputs for FX8010 (see below) extout diff --git a/Documentation/sound/cards/audigy-mixer.rst b/Documentation/sound/cards/audigy-mixer.rst index c506f8d16f2e..aa176451d5b5 100644 --- a/Documentation/sound/cards/audigy-mixer.rst +++ b/Documentation/sound/cards/audigy-mixer.rst @@ -19,9 +19,9 @@ Digital mixer controls These controls are built using the DSP instructions. They offer extended functionality. Only the default built-in code in the ALSA driver is described here. Note that the controls work as attenuators: the maximum value is the -neutral position leaving the signal unchanged. Note that if the same destination -is mentioned in multiple controls, the signal is accumulated and can be wrapped -(set to maximal or minimal value without checking of overflow). +neutral position leaving the signal unchanged. Note that if the same destination +is mentioned in multiple controls, the signal is accumulated and can be clipped +(set to maximal or minimal value without checking for overflow). Explanation of used abbreviations: @@ -32,17 +32,17 @@ ADC analog to digital converter I2S one-way three wire serial bus for digital sound by Philips Semiconductors - (this standard is used for connecting standalone DAC and ADC converters) + (this standard is used for connecting standalone D/A and A/D converters) LFE - low frequency effects (subwoofer signal) + low frequency effects (used as subwoofer signal) AC97 - a chip containing an analog mixer, DAC and ADC converters + a chip containing an analog mixer, D/A and A/D converters IEC958 S/PDIF FX-bus the EMU10K2 chip has an effect bus containing 64 accumulators. - Each of the synthesizer voices can feed its output to these accumulators - and the DSP microcontroller can operate with the resulting sum. + Each of the synthesizer voices can feed its output to these accumulators + and the DSP microcontroller can operate with the resulting sum. name='PCM Front Playback Volume',index=0 ---------------------------------------- @@ -218,8 +218,8 @@ LFE outputs. name='IEC958 Optical Raw Playback Switch',index=0 ------------------------------------------------- If this switch is on, then the samples for the IEC958 (S/PDIF) digital -output are taken only from the raw FX8010 PCM, otherwise standard front -PCM samples are taken. +output are taken only from the raw iec958 ALSA PCM device (which uses +accumulators 20 and 21 for left and right PCM by default). PCM stream related controls @@ -237,8 +237,8 @@ as follows: name='EMU10K1 PCM Send Routing',index 0-31 ------------------------------------------ -This control specifies the destination - FX-bus accumulators. There 24 -values with this mapping: +This control specifies the destination - FX-bus accumulators. There are 24 +values in this mapping: * 0 - mono, A destination (FX-bus 0-63), default 0 * 1 - mono, B destination (FX-bus 0-63), default 1 @@ -306,6 +306,9 @@ MANUALS/PATENTS ftp://opensource.creative.com/pub/doc ------------------------------------- +Note that the site is defunct, but the documents are available +from various other locations. + LM4545.pdf AC97 Codec diff --git a/Documentation/sound/cards/sb-live-mixer.rst b/Documentation/sound/cards/sb-live-mixer.rst index 357fcd619d39..819886634400 100644 --- a/Documentation/sound/cards/sb-live-mixer.rst +++ b/Documentation/sound/cards/sb-live-mixer.rst @@ -15,7 +15,7 @@ The ALSA driver programs this portion of chip by default code IEC958 (S/PDIF) raw PCM ======================= -This PCM device (it's the 4th PCM device (index 3!) and first subdevice +This PCM device (it's the 3rd PCM device (index 2!) and first subdevice (index 0) for a given card) allows to forward 48kHz, stereo, 16-bit little endian streams without any modifications to the digital output (coaxial or optical). The universal interface allows the creation of up @@ -33,9 +33,9 @@ Digital mixer controls These controls are built using the DSP instructions. They offer extended functionality. Only the default built-in code in the ALSA driver is described here. Note that the controls work as attenuators: the maximum value is the -neutral position leaving the signal unchanged. Note that if the same destination -is mentioned in multiple controls, the signal is accumulated and can be wrapped -(set to maximal or minimal value without checking of overflow). +neutral position leaving the signal unchanged. Note that if the same destination +is mentioned in multiple controls, the signal is accumulated and can be clipped +(set to maximal or minimal value without checking for overflow). Explanation of used abbreviations: @@ -46,11 +46,11 @@ ADC analog to digital converter I2S one-way three wire serial bus for digital sound by Philips Semiconductors - (this standard is used for connecting standalone DAC and ADC converters) + (this standard is used for connecting standalone D/A and A/D converters) LFE - low frequency effects (subwoofer signal) + low frequency effects (used as subwoofer signal) AC97 - a chip containing an analog mixer, DAC and ADC converters + a chip containing an analog mixer, D/A and A/D converters IEC958 S/PDIF FX-bus @@ -313,6 +313,9 @@ MANUALS/PATENTS ftp://opensource.creative.com/pub/doc ------------------------------------- +Note that the site is defunct, but the documents are available +from various other locations. + LM4545.pdf AC97 Codec m2049.pdf diff --git a/Documentation/sound/hd-audio/models.rst b/Documentation/sound/hd-audio/models.rst index 9b52f50a6854..120430450014 100644 --- a/Documentation/sound/hd-audio/models.rst +++ b/Documentation/sound/hd-audio/models.rst @@ -704,7 +704,7 @@ ref no-jd BIOS setup but without jack-detection intel - Intel DG45* mobos + Intel D*45* mobos dell-m6-amic Dell desktops/laptops with analog mics dell-m6-dmic diff --git a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst index 5c9523b7d55c..4335c98b3d82 100644 --- a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst +++ b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst @@ -19,18 +19,13 @@ explain the general topic of linux kernel coding and doesn't cover low-level driver implementation details. It only describes the standard way to write a PCI sound driver on ALSA. -This document is still a draft version. Any feedback and corrections, -please!! - File Tree Structure =================== General ------- -The file tree structure of ALSA driver is depicted below. - -:: +The file tree structure of ALSA driver is depicted below:: sound /core @@ -68,8 +63,8 @@ kernel config. core/oss ~~~~~~~~ -The codes for PCM and mixer OSS emulation modules are stored in this -directory. The rawmidi OSS emulation is included in the ALSA rawmidi +The code for OSS PCM and mixer emulation modules is stored in this +directory. The OSS rawmidi emulation is included in the ALSA rawmidi code since it's quite small. The sequencer code is stored in ``core/seq/oss`` directory (see `below <core/seq/oss_>`__). @@ -78,19 +73,19 @@ core/seq This directory and its sub-directories are for the ALSA sequencer. This directory contains the sequencer core and primary sequencer modules such -like snd-seq-midi, snd-seq-virmidi, etc. They are compiled only when +as snd-seq-midi, snd-seq-virmidi, etc. They are compiled only when ``CONFIG_SND_SEQUENCER`` is set in the kernel config. core/seq/oss ~~~~~~~~~~~~ -This contains the OSS sequencer emulation codes. +This contains the OSS sequencer emulation code. include directory ----------------- This is the place for the public header files of ALSA drivers, which are -to be exported to user-space, or included by several files at different +to be exported to user-space, or included by several files in different directories. Basically, the private header files should not be placed in this directory, but you may still find files there, due to historical reasons :) @@ -100,7 +95,7 @@ drivers directory This directory contains code shared among different drivers on different architectures. They are hence supposed not to be architecture-specific. -For example, the dummy pcm driver and the serial MIDI driver are found +For example, the dummy PCM driver and the serial MIDI driver are found in this directory. In the sub-directories, there is code for components which are independent from bus and cpu architectures. @@ -156,8 +151,8 @@ these architectures. usb directory ------------- -This directory contains the USB-audio driver. In the latest version, the -USB MIDI driver is integrated in the usb-audio driver. +This directory contains the USB-audio driver. +The USB MIDI driver is integrated in the usb-audio driver. pcmcia directory ---------------- @@ -175,9 +170,9 @@ layer including ASoC core, codec and machine drivers. oss directory ------------- -Here contains OSS/Lite codes. -All codes have been deprecated except for dmasound on m68k as of -writing this. +This contains OSS/Lite code. +At the time of writing, all code has been removed except for dmasound +on m68k. Basic Flow for PCI Drivers @@ -341,7 +336,7 @@ to details explained in the following section. error: snd_card_free(card); - return err; + return err; } /* destructor -- see the "Destructor" sub-section */ @@ -381,7 +376,7 @@ where ``enable[dev]`` is the module option. Each time the ``probe`` callback is called, check the availability of the device. If not available, simply increment the device index and -returns. dev will be incremented also later (`step 7 +return. dev will be incremented also later (`step 7 <7) Set the PCI driver data and return zero._>`__). 2) Create a card instance @@ -402,9 +397,7 @@ Components`_. 3) Create a main component ~~~~~~~~~~~~~~~~~~~~~~~~~~ -In this part, the PCI resources are allocated. - -:: +In this part, the PCI resources are allocated:: struct mychip *chip; .... @@ -417,13 +410,11 @@ Management`_. When something goes wrong, the probe function needs to deal with the error. In this example, we have a single error handling path placed -at the end of the function. - -:: +at the end of the function:: error: snd_card_free(card); - return err; + return err; Since each component can be properly freed, the single :c:func:`snd_card_free()` call should suffice in most cases. @@ -483,13 +474,11 @@ remove callback and power-management callbacks, too. Destructor ---------- -The destructor, remove callback, simply releases the card instance. Then -the ALSA middle layer will release all the attached components +The destructor, the remove callback, simply releases the card instance. +Then the ALSA middle layer will release all the attached components automatically. -It would be typically just calling :c:func:`snd_card_free()`: - -:: +It would be typically just calling :c:func:`snd_card_free()`:: static void snd_mychip_remove(struct pci_dev *pci) { @@ -504,9 +493,7 @@ Header Files ------------ For the above example, at least the following include files are -necessary. - -:: +necessary:: #include <linux/init.h> #include <linux/pci.h> @@ -544,9 +531,7 @@ list on the card record is used to manage the correct release of resources at destruction. As mentioned above, to create a card instance, call -:c:func:`snd_card_new()`. - -:: +:c:func:`snd_card_new()`:: struct snd_card *card; int err; @@ -572,10 +557,8 @@ struct snd_device object. A component can be a PCM instance, a control interface, a raw MIDI interface, etc. Each such instance has one component entry. -A component can be created via :c:func:`snd_device_new()` -function. - -:: +A component can be created via the :c:func:`snd_device_new()` +function:: snd_device_new(card, SNDRV_DEV_XXX, chip, &ops); @@ -591,7 +574,7 @@ allocated manually beforehand, and its pointer is passed as the argument. This pointer (``chip`` in the above example) is used as the identifier for the instance. -Each pre-defined ALSA component such as ac97 and pcm calls +Each pre-defined ALSA component such as AC97 and PCM calls :c:func:`snd_device_new()` inside its constructor. The destructor for each component is defined in the callback pointers. Hence, you don't need to take care of calling a destructor for such a component. @@ -605,9 +588,7 @@ Chip-Specific Data ------------------ Chip-specific information, e.g. the I/O port address, its resource -pointer, or the irq number, is stored in the chip-specific record. - -:: +pointer, or the irq number, is stored in the chip-specific record:: struct mychip { .... @@ -620,9 +601,7 @@ In general, there are two ways of allocating the chip record. ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ As mentioned above, you can pass the extra-data-length to the 5th -argument of :c:func:`snd_card_new()`, i.e. - -:: +argument of :c:func:`snd_card_new()`, e.g.:: err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, sizeof(struct mychip), &card); @@ -642,9 +621,7 @@ released together with the card instance. ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ After allocating a card instance via :c:func:`snd_card_new()` -(with ``0`` on the 4th arg), call :c:func:`kzalloc()`. - -:: +(with ``0`` on the 4th arg), call :c:func:`kzalloc()`:: struct snd_card *card; struct mychip *chip; @@ -663,16 +640,12 @@ The chip record should have the field to hold the card pointer at least, }; -Then, set the card pointer in the returned chip instance. - -:: +Then, set the card pointer in the returned chip instance:: chip->card = card; Next, initialize the fields, and register this chip record as a -low-level device with a specified ``ops``, - -:: +low-level device with a specified ``ops``:: static const struct snd_device_ops ops = { .dev_free = snd_mychip_dev_free, @@ -681,9 +654,7 @@ low-level device with a specified ``ops``, snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); :c:func:`snd_mychip_dev_free()` is the device-destructor -function, which will call the real destructor. - -:: +function, which will call the real destructor:: static int snd_mychip_dev_free(struct snd_device *device) { @@ -692,10 +663,10 @@ function, which will call the real destructor. where :c:func:`snd_mychip_free()` is the real destructor. -The demerit of this method is the obviously more amount of codes. -The merit is, however, you can trigger the own callback at registering -and disconnecting the card via setting in snd_device_ops. -About the registering and disconnecting the card, see the subsections +The demerit of this method is the obviously larger amount of code. +The merit is, however, that you can trigger your own callback at +registering and disconnecting the card via a setting in snd_device_ops. +About registering and disconnecting the card, see the subsections below. @@ -724,9 +695,7 @@ Full Code Example ----------------- In this section, we'll complete the chip-specific constructor, -destructor and PCI entries. Example code is shown first, below. - -:: +destructor and PCI entries. Example code is shown first, below:: struct mychip { struct snd_card *card; @@ -866,9 +835,7 @@ resources. Also, you need to set the proper PCI DMA mask to limit the accessed I/O range. In some cases, you might need to call :c:func:`pci_set_master()` function, too. -Suppose the 28bit mask, and the code to be added would be like: - -:: +Suppose a 28bit mask, the code to be added would look like:: err = pci_enable_device(pci); if (err < 0) @@ -890,9 +857,7 @@ function (see below). Now assume that the PCI device has an I/O port with 8 bytes and an interrupt. Then struct mychip will have the -following fields: - -:: +following fields:: struct mychip { struct snd_card *card; @@ -905,14 +870,12 @@ following fields: For an I/O port (and also a memory region), you need to have the resource pointer for the standard resource management. For an irq, you have to keep only the irq number (integer). But you need to initialize -this number as -1 before actual allocation, since irq 0 is valid. The +this number to -1 before actual allocation, since irq 0 is valid. The port address and its resource pointer can be initialized as null by :c:func:`kzalloc()` automatically, so you don't have to take care of resetting them. -The allocation of an I/O port is done like this: - -:: +The allocation of an I/O port is done like this:: err = pci_request_regions(pci, "My Chip"); if (err < 0) { @@ -928,9 +891,7 @@ The returned value, ``chip->res_port``, is allocated via must be released via :c:func:`kfree()`, but there is a problem with this. This issue will be explained later. -The allocation of an interrupt source is done like this: - -:: +The allocation of an interrupt source is done like this:: if (request_irq(pci->irq, snd_mychip_interrupt, IRQF_SHARED, KBUILD_MODNAME, chip)) { @@ -954,9 +915,7 @@ used for that, but you can use what you like, too. I won't give details about the interrupt handler at this point, but at least its appearance can be explained now. The interrupt handler looks -usually like the following: - -:: +usually as follows:: static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id) { @@ -966,13 +925,12 @@ usually like the following: } After requesting the IRQ, you can passed it to ``card->sync_irq`` -field: -:: +field:: card->irq = chip->irq; -This allows PCM core automatically performing -:c:func:`synchronize_irq()` at the necessary timing like ``hw_free``. +This allows the PCM core to automatically call +:c:func:`synchronize_irq()` at the right time, like before ``hw_free``. See the later section `sync_stop callback`_ for details. Now let's write the corresponding destructor for the resources above. @@ -981,9 +939,7 @@ activated) and release the resources. So far, we have no hardware part, so the disabling code is not written here. To release the resources, the “check-and-release” method is a safer way. -For the interrupt, do like this: - -:: +For the interrupt, do like this:: if (chip->irq >= 0) free_irq(chip->irq, chip); @@ -997,9 +953,7 @@ When you requested I/O ports or memory regions via :c:func:`pci_request_regions()` like in this example, release the resource(s) using the corresponding function, :c:func:`pci_release_region()` or -:c:func:`pci_release_regions()`. - -:: +:c:func:`pci_release_regions()`:: pci_release_regions(chip->pci); @@ -1007,39 +961,32 @@ When you requested manually via :c:func:`request_region()` or :c:func:`request_mem_region()`, you can release it via :c:func:`release_resource()`. Suppose that you keep the resource pointer returned from :c:func:`request_region()` in -chip->res_port, the release procedure looks like: - -:: +chip->res_port, the release procedure looks like:: release_and_free_resource(chip->res_port); Don't forget to call :c:func:`pci_disable_device()` before the end. -And finally, release the chip-specific record. - -:: +And finally, release the chip-specific record:: kfree(chip); -We didn't implement the hardware disabling part in the above. If you +We didn't implement the hardware disabling part above. If you need to do this, please note that the destructor may be called even before the initialization of the chip is completed. It would be better to have a flag to skip hardware disabling if the hardware was not initialized yet. When the chip-data is assigned to the card using -:c:func:`snd_device_new()` with ``SNDRV_DEV_LOWLELVEL`` , its -destructor is called at the last. That is, it is assured that all other +:c:func:`snd_device_new()` with ``SNDRV_DEV_LOWLELVEL``, its +destructor is called last. That is, it is assured that all other components like PCMs and controls have already been released. You don't have to stop PCMs, etc. explicitly, but just call low-level hardware stopping. The management of a memory-mapped region is almost as same as the -management of an I/O port. You'll need three fields like the -following: - -:: +management of an I/O port. You'll need two fields as follows:: struct mychip { .... @@ -1047,9 +994,7 @@ following: void __iomem *iobase_virt; }; -and the allocation would be like below: - -:: +and the allocation would look like below:: err = pci_request_regions(pci, "My Chip"); if (err < 0) { @@ -1060,9 +1005,7 @@ and the allocation would be like below: chip->iobase_virt = ioremap(chip->iobase_phys, pci_resource_len(pci, 0)); -and the corresponding destructor would be: - -:: +and the corresponding destructor would be:: static int snd_mychip_free(struct mychip *chip) { @@ -1075,9 +1018,7 @@ and the corresponding destructor would be: } Of course, a modern way with :c:func:`pci_iomap()` will make things a -bit easier, too. - -:: +bit easier, too:: err = pci_request_regions(pci, "My Chip"); if (err < 0) { @@ -1097,9 +1038,7 @@ struct pci_device_id table for this chipset. It's a table of PCI vendor/device ID number, and some masks. -For example, - -:: +For example:: static struct pci_device_id snd_mychip_ids[] = { { PCI_VENDOR_ID_FOO, PCI_DEVICE_ID_BAR, @@ -1120,9 +1059,7 @@ The last entry of this list is the terminator. You must specify this all-zero entry. Then, prepare the struct pci_driver -record: - -:: +record:: static struct pci_driver driver = { .name = KBUILD_MODNAME, @@ -1133,11 +1070,9 @@ record: The ``probe`` and ``remove`` functions have already been defined in the previous sections. The ``name`` field is the name string of this -device. Note that you must not use a slash “/” in this string. - -And at last, the module entries: +device. Note that you must not use slashes (“/”) in this string. -:: +And at last, the module entries:: static int __init alsa_card_mychip_init(void) { @@ -1167,22 +1102,22 @@ The PCM middle layer of ALSA is quite powerful and it is only necessary for each driver to implement the low-level functions to access its hardware. -For accessing to the PCM layer, you need to include ``<sound/pcm.h>`` +To access the PCM layer, you need to include ``<sound/pcm.h>`` first. In addition, ``<sound/pcm_params.h>`` might be needed if you -access to some functions related with hw_param. +access some functions related with hw_param. -Each card device can have up to four pcm instances. A pcm instance -corresponds to a pcm device file. The limitation of number of instances -comes only from the available bit size of the Linux's device numbers. -Once when 64bit device number is used, we'll have more pcm instances +Each card device can have up to four PCM instances. A PCM instance +corresponds to a PCM device file. The limitation of number of instances +comes only from the available bit size of Linux' device numbers. +Once 64bit device numbers are used, we'll have more PCM instances available. -A pcm instance consists of pcm playback and capture streams, and each -pcm stream consists of one or more pcm substreams. Some soundcards +A PCM instance consists of PCM playback and capture streams, and each +PCM stream consists of one or more PCM substreams. Some soundcards support multiple playback functions. For example, emu10k1 has a PCM playback of 32 stereo substreams. In this case, at each open, a free substream is (usually) automatically chosen and opened. Meanwhile, when -only one substream exists and it was already opened, the successful open +only one substream exists and it was already opened, a subsequent open will either block or error with ``EAGAIN`` according to the file open mode. But you don't have to care about such details in your driver. The PCM middle layer will take care of such work. @@ -1191,9 +1126,7 @@ Full Code Example ----------------- The example code below does not include any hardware access routines but -shows only the skeleton, how to build up the PCM interfaces. - -:: +shows only the skeleton, how to build up the PCM interfaces:: #include <sound/pcm.h> .... @@ -1399,10 +1332,8 @@ shows only the skeleton, how to build up the PCM interfaces. PCM Constructor --------------- -A pcm instance is allocated by the :c:func:`snd_pcm_new()` -function. It would be better to create a constructor for pcm, namely, - -:: +A PCM instance is allocated by the :c:func:`snd_pcm_new()` +function. It would be better to create a constructor for the PCM, namely:: static int snd_mychip_new_pcm(struct mychip *chip) { @@ -1415,16 +1346,16 @@ function. It would be better to create a constructor for pcm, namely, pcm->private_data = chip; strcpy(pcm->name, "My Chip"); chip->pcm = pcm; - .... + ... return 0; } -The :c:func:`snd_pcm_new()` function takes four arguments. The -first argument is the card pointer to which this pcm is assigned, and +The :c:func:`snd_pcm_new()` function takes six arguments. The +first argument is the card pointer to which this PCM is assigned, and the second is the ID string. The third argument (``index``, 0 in the above) is the index of this new -pcm. It begins from zero. If you create more than one pcm instances, +PCM. It begins from zero. If you create more than one PCM instances, specify the different numbers in this argument. For example, ``index = 1`` for the second PCM device. @@ -1437,26 +1368,20 @@ If a chip supports multiple playbacks or captures, you can specify more numbers, but they must be handled properly in open/close, etc. callbacks. When you need to know which substream you are referring to, then it can be obtained from struct snd_pcm_substream data passed to each -callback as follows: - -:: +callback as follows:: struct snd_pcm_substream *substream; int index = substream->number; -After the pcm is created, you need to set operators for each pcm stream. - -:: +After the PCM is created, you need to set operators for each PCM stream:: snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_mychip_playback_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_mychip_capture_ops); -The operators are defined typically like this: - -:: +The operators are defined typically like this:: static struct snd_pcm_ops snd_mychip_playback_ops = { .open = snd_mychip_pcm_open, @@ -1472,25 +1397,21 @@ All the callbacks are described in the Operators_ subsection. After setting the operators, you probably will want to pre-allocate the buffer and set up the managed allocation mode. -For that, simply call the following: - -:: +For that, simply call the following:: snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV, &chip->pci->dev, 64*1024, 64*1024); -It will allocate a buffer up to 64kB as default. Buffer management +It will allocate a buffer up to 64kB by default. Buffer management details will be described in the later section `Buffer and Memory Management`_. -Additionally, you can set some extra information for this pcm in +Additionally, you can set some extra information for this PCM in ``pcm->info_flags``. The available values are defined as ``SNDRV_PCM_INFO_XXX`` in ``<sound/asound.h>``, which is used for the hardware definition (described later). When your soundchip supports only -half-duplex, specify like this: - -:: +half-duplex, specify it like this:: pcm->info_flags = SNDRV_PCM_INFO_HALF_DUPLEX; @@ -1498,15 +1419,13 @@ half-duplex, specify like this: ... And the Destructor? ----------------------- -The destructor for a pcm instance is not always necessary. Since the pcm +The destructor for a PCM instance is not always necessary. Since the PCM device will be released by the middle layer code automatically, you don't have to call the destructor explicitly. The destructor would be necessary if you created special records internally and needed to release them. In such a case, set the -destructor function to ``pcm->private_free``: - -:: +destructor function to ``pcm->private_free``:: static void mychip_pcm_free(struct snd_pcm *pcm) { @@ -1537,13 +1456,11 @@ Runtime Pointer - The Chest of PCM Information When the PCM substream is opened, a PCM runtime instance is allocated and assigned to the substream. This pointer is accessible via ``substream->runtime``. This runtime pointer holds most information you -need to control the PCM: the copy of hw_params and sw_params +need to control the PCM: a copy of hw_params and sw_params configurations, the buffer pointers, mmap records, spinlocks, etc. The definition of runtime instance is found in ``<sound/pcm.h>``. Here -are the contents of this file: - -:: +is the relevant part of this file:: struct _snd_pcm_runtime { /* -- Status -- */ @@ -1577,14 +1494,19 @@ are the contents of this file: unsigned int period_step; unsigned int sleep_min; /* min ticks to sleep */ snd_pcm_uframes_t start_threshold; - snd_pcm_uframes_t stop_threshold; - snd_pcm_uframes_t silence_threshold; /* Silence filling happens when - noise is nearest than this */ - snd_pcm_uframes_t silence_size; /* Silence filling size */ + /* + * The following two thresholds alleviate playback buffer underruns; when + * hw_avail drops below the threshold, the respective action is triggered: + */ + snd_pcm_uframes_t stop_threshold; /* - stop playback */ + snd_pcm_uframes_t silence_threshold; /* - pre-fill buffer with silence */ + snd_pcm_uframes_t silence_size; /* max size of silence pre-fill; when >= boundary, + * fill played area with silence immediately */ snd_pcm_uframes_t boundary; /* pointers wrap point */ - snd_pcm_uframes_t silenced_start; - snd_pcm_uframes_t silenced_size; + /* internal data of auto-silencer */ + snd_pcm_uframes_t silence_start; /* starting pointer to silence area */ + snd_pcm_uframes_t silence_filled; /* size filled with silence */ snd_pcm_sync_id_t sync; /* hardware synchronization ID */ @@ -1638,14 +1560,12 @@ Hardware Description The hardware descriptor (struct snd_pcm_hardware) contains the definitions of the fundamental hardware configuration. Above all, you'll need to define this -in the `PCM open callback`_. Note that the runtime instance holds the copy of -the descriptor, not the pointer to the existing descriptor. That is, +in the `PCM open callback`_. Note that the runtime instance holds a copy of +the descriptor, not a pointer to the existing descriptor. That is, in the open callback, you can modify the copied descriptor (``runtime->hw``) as you need. For example, if the maximum number of channels is 1 only on some chip models, you can still use the same -hardware descriptor and change the channels_max later: - -:: +hardware descriptor and change the channels_max later:: struct snd_pcm_runtime *runtime = substream->runtime; ... @@ -1653,9 +1573,7 @@ hardware descriptor and change the channels_max later: if (chip->model == VERY_OLD_ONE) runtime->hw.channels_max = 1; -Typically, you'll have a hardware descriptor as below: - -:: +Typically, you'll have a hardware descriptor as below:: static struct snd_pcm_hardware snd_mychip_playback_hw = { .info = (SNDRV_PCM_INFO_MMAP | @@ -1676,51 +1594,51 @@ Typically, you'll have a hardware descriptor as below: }; - The ``info`` field contains the type and capabilities of this - pcm. The bit flags are defined in ``<sound/asound.h>`` as + PCM. The bit flags are defined in ``<sound/asound.h>`` as ``SNDRV_PCM_INFO_XXX``. Here, at least, you have to specify whether - the mmap is supported and which interleaved format is + mmap is supported and which interleaving formats are supported. When the hardware supports mmap, add the ``SNDRV_PCM_INFO_MMAP`` flag here. When the hardware supports the - interleaved or the non-interleaved formats, + interleaved or the non-interleaved formats, the ``SNDRV_PCM_INFO_INTERLEAVED`` or ``SNDRV_PCM_INFO_NONINTERLEAVED`` flag must be set, respectively. If both are supported, you can set both, too. In the above example, ``MMAP_VALID`` and ``BLOCK_TRANSFER`` are specified for the OSS mmap mode. Usually both are set. Of course, - ``MMAP_VALID`` is set only if the mmap is really supported. + ``MMAP_VALID`` is set only if mmap is really supported. The other possible flags are ``SNDRV_PCM_INFO_PAUSE`` and - ``SNDRV_PCM_INFO_RESUME``. The ``PAUSE`` bit means that the pcm + ``SNDRV_PCM_INFO_RESUME``. The ``PAUSE`` bit means that the PCM supports the “pause” operation, while the ``RESUME`` bit means that - the pcm supports the full “suspend/resume” operation. If the + the PCM supports the full “suspend/resume” operation. If the ``PAUSE`` flag is set, the ``trigger`` callback below must handle the corresponding (pause push/release) commands. The suspend/resume trigger commands can be defined even without the ``RESUME`` - flag. See `Power Management`_ section for details. + flag. See the `Power Management`_ section for details. When the PCM substreams can be synchronized (typically, - synchronized start/stop of a playback and a capture streams), you + synchronized start/stop of a playback and a capture stream), you can give ``SNDRV_PCM_INFO_SYNC_START``, too. In this case, you'll need to check the linked-list of PCM substreams in the trigger - callback. This will be described in the later section. + callback. This will be described in a later section. -- ``formats`` field contains the bit-flags of supported formats +- The ``formats`` field contains the bit-flags of supported formats (``SNDRV_PCM_FMTBIT_XXX``). If the hardware supports more than one format, give all or'ed bits. In the example above, the signed 16bit little-endian format is specified. -- ``rates`` field contains the bit-flags of supported rates +- The ``rates`` field contains the bit-flags of supported rates (``SNDRV_PCM_RATE_XXX``). When the chip supports continuous rates, - pass ``CONTINUOUS`` bit additionally. The pre-defined rate bits are - provided only for typical rates. If your chip supports + pass the ``CONTINUOUS`` bit additionally. The pre-defined rate bits + are provided only for typical rates. If your chip supports unconventional rates, you need to add the ``KNOT`` bit and set up the hardware constraint manually (explained later). - ``rate_min`` and ``rate_max`` define the minimum and maximum sample rate. This should correspond somehow to ``rates`` bits. -- ``channels_min`` and ``channels_max`` define, as you might already +- ``channels_min`` and ``channels_max`` define, as you might have already expected, the minimum and maximum number of channels. - ``buffer_bytes_max`` defines the maximum buffer size in @@ -1732,15 +1650,16 @@ Typically, you'll have a hardware descriptor as below: number of periods in the buffer. The “period” is a term that corresponds to a fragment in the OSS - world. The period defines the size at which a PCM interrupt is - generated. This size strongly depends on the hardware. Generally, - the smaller period size will give you more interrupts, that is, - more controls. In the case of capture, this size defines the input - latency. On the other hand, the whole buffer size defines the - output latency for the playback direction. + world. The period defines the point at which a PCM interrupt is + generated. This point strongly depends on the hardware. Generally, + a smaller period size will give you more interrupts, which results + in being able to fill/drain the buffer more timely. In the case of + capture, this size defines the input latency. On the other hand, + the whole buffer size defines the output latency for the playback + direction. - There is also a field ``fifo_size``. This specifies the size of the - hardware FIFO, but currently it is neither used in the driver nor + hardware FIFO, but currently it is neither used by the drivers nor in the alsa-lib. So, you can ignore this field. PCM Configurations @@ -1759,34 +1678,32 @@ One thing to be noted is that the configured buffer and period sizes are stored in “frames” in the runtime. In the ALSA world, ``1 frame = channels \* samples-size``. For conversion between frames and bytes, you can use the :c:func:`frames_to_bytes()` and -:c:func:`bytes_to_frames()` helper functions. - -:: +:c:func:`bytes_to_frames()` helper functions:: period_bytes = frames_to_bytes(runtime, runtime->period_size); Also, many software parameters (sw_params) are stored in frames, too. -Please check the type of the field. ``snd_pcm_uframes_t`` is for the -frames as unsigned integer while ``snd_pcm_sframes_t`` is for the +Please check the type of the field. ``snd_pcm_uframes_t`` is for +frames as unsigned integer while ``snd_pcm_sframes_t`` is for frames as signed integer. DMA Buffer Information ~~~~~~~~~~~~~~~~~~~~~~ -The DMA buffer is defined by the following four fields, ``dma_area``, -``dma_addr``, ``dma_bytes`` and ``dma_private``. The ``dma_area`` +The DMA buffer is defined by the following four fields: ``dma_area``, +``dma_addr``, ``dma_bytes`` and ``dma_private``. ``dma_area`` holds the buffer pointer (the logical address). You can call :c:func:`memcpy()` from/to this pointer. Meanwhile, ``dma_addr`` holds the physical address of the buffer. This field is specified only when -the buffer is a linear buffer. ``dma_bytes`` holds the size of buffer -in bytes. ``dma_private`` is used for the ALSA DMA allocator. +the buffer is a linear buffer. ``dma_bytes`` holds the size of the +buffer in bytes. ``dma_private`` is used for the ALSA DMA allocator. If you use either the managed buffer allocation mode or the standard API function :c:func:`snd_pcm_lib_malloc_pages()` for allocating the buffer, these fields are set by the ALSA middle layer, and you should *not* change them by yourself. You can read them but not write them. On the other hand, if you want to allocate the buffer by yourself, you'll -need to manage it in hw_params callback. At least, ``dma_bytes`` is +need to manage it in the hw_params callback. At least, ``dma_bytes`` is mandatory. ``dma_area`` is necessary when the buffer is mmapped. If your driver doesn't support mmap, this field is not necessary. ``dma_addr`` is also optional. You can use dma_private as @@ -1796,13 +1713,13 @@ Running Status ~~~~~~~~~~~~~~ The running status can be referred via ``runtime->status``. This is -the pointer to the struct snd_pcm_mmap_status record. +a pointer to a struct snd_pcm_mmap_status record. For example, you can get the current DMA hardware pointer via ``runtime->status->hw_ptr``. The DMA application pointer can be referred via ``runtime->control``, -which points to the struct snd_pcm_mmap_control record. -However, accessing directly to this value is not recommended. +which points to a struct snd_pcm_mmap_control record. +However, accessing this value directly is not recommended. Private Data ~~~~~~~~~~~~ @@ -1811,11 +1728,10 @@ You can allocate a record for the substream and store it in ``runtime->private_data``. Usually, this is done in the `PCM open callback`_. Don't mix this with ``pcm->private_data``. The ``pcm->private_data`` usually points to the chip instance assigned -statically at the creation of PCM, while the ``runtime->private_data`` -points to a dynamic data structure created at the PCM open -callback. - -:: +statically at creation time of the PCM device, while +``runtime->private_data`` +points to a dynamic data structure created in the PCM open +callback:: static int snd_xxx_open(struct snd_pcm_substream *substream) { @@ -1832,20 +1748,18 @@ The allocated object must be released in the `close callback`_. Operators --------- -OK, now let me give details about each pcm callback (``ops``). In +OK, now let me give details about each PCM callback (``ops``). In general, every callback must return 0 if successful, or a negative error number such as ``-EINVAL``. To choose an appropriate error number, it is advised to check what value other parts of the kernel return when the same kind of request fails. -The callback function takes at least the argument with +Each callback function takes at least one argument containing a struct snd_pcm_substream pointer. To retrieve the chip record from the given substream instance, you can use the following -macro. - -:: +macro:: - int xxx() { + int xxx(...) { struct mychip *chip = snd_pcm_substream_chip(substream); .... } @@ -1864,12 +1778,10 @@ PCM open callback static int snd_xxx_open(struct snd_pcm_substream *substream); -This is called when a pcm substream is opened. +This is called when a PCM substream is opened. At least, here you have to initialize the ``runtime->hw`` -record. Typically, this is done by like this: - -:: +record. Typically, this is done like this:: static int snd_xxx_open(struct snd_pcm_substream *substream) { @@ -1883,7 +1795,7 @@ record. Typically, this is done by like this: where ``snd_mychip_playback_hw`` is the pre-defined hardware description. -You can allocate a private data in this callback, as described in +You can allocate private data in this callback, as described in the `Private Data`_ section. If the hardware configuration needs more constraints, set the hardware @@ -1897,12 +1809,10 @@ close callback static int snd_xxx_close(struct snd_pcm_substream *substream); -Obviously, this is called when a pcm substream is closed. - -Any private instance for a pcm substream allocated in the ``open`` -callback will be released here. +Obviously, this is called when a PCM substream is closed. -:: +Any private instance for a PCM substream allocated in the ``open`` +callback will be released here:: static int snd_xxx_close(struct snd_pcm_substream *substream) { @@ -1914,9 +1824,9 @@ callback will be released here. ioctl callback ~~~~~~~~~~~~~~ -This is used for any special call to pcm ioctls. But usually you can -leave it as NULL, then PCM core calls the generic ioctl callback -function :c:func:`snd_pcm_lib_ioctl()`. If you need to deal with the +This is used for any special call to PCM ioctls. But usually you can +leave it NULL, then the PCM core calls the generic ioctl callback +function :c:func:`snd_pcm_lib_ioctl()`. If you need to deal with a unique setup of channel info or reset procedure, you can pass your own callback function here. @@ -1928,22 +1838,20 @@ hw_params callback static int snd_xxx_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params); -This is called when the hardware parameter (``hw_params``) is set up +This is called when the hardware parameters (``hw_params``) are set up by the application, that is, once when the buffer size, the period -size, the format, etc. are defined for the pcm substream. +size, the format, etc. are defined for the PCM substream. Many hardware setups should be done in this callback, including the allocation of buffers. -Parameters to be initialized are retrieved by +Parameters to be initialized are retrieved by the :c:func:`params_xxx()` macros. -When you set up the managed buffer allocation mode for the substream, +When you choose managed buffer allocation mode for the substream, a buffer is already allocated before this callback gets called. Alternatively, you can call a helper function below for -allocating the buffer, too. - -:: +allocating the buffer:: snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); @@ -1951,8 +1859,8 @@ allocating the buffer, too. DMA buffers have been pre-allocated. See the section `Buffer Types`_ for more details. -Note that this and ``prepare`` callbacks may be called multiple times -per initialization. For example, the OSS emulation may call these +Note that this one and the ``prepare`` callback may be called multiple +times per initialization. For example, the OSS emulation may call these callbacks at each change via its ioctl. Thus, you need to be careful not to allocate the same buffers many @@ -1960,10 +1868,10 @@ times, which will lead to memory leaks! Calling the helper function above many times is OK. It will release the previous buffer automatically when it was already allocated. -Another note is that this callback is non-atomic (schedulable) as +Another note is that this callback is non-atomic (schedulable) by default, i.e. when no ``nonatomic`` flag set. This is important, because the ``trigger`` callback is atomic (non-schedulable). That is, -mutexes or any schedule-related functions are not available in +mutexes or any schedule-related functions are not available in the ``trigger`` callback. Please see the subsection Atomicity_ for details. @@ -1979,16 +1887,14 @@ This is called to release the resources allocated via This function is always called before the close callback is called. Also, the callback may be called multiple times, too. Keep track -whether the resource was already released. +whether each resource was already released. -When you have set up the managed buffer allocation mode for the PCM +When you have chosen managed buffer allocation mode for the PCM substream, the allocated PCM buffer will be automatically released after this callback gets called. Otherwise you'll have to release the buffer manually. Typically, when the buffer was allocated from the pre-allocated pool, you can use the standard API function -:c:func:`snd_pcm_lib_malloc_pages()` like: - -:: +:c:func:`snd_pcm_lib_malloc_pages()` like:: snd_pcm_lib_free_pages(substream); @@ -1999,13 +1905,13 @@ prepare callback static int snd_xxx_prepare(struct snd_pcm_substream *substream); -This callback is called when the pcm is “prepared”. You can set the +This callback is called when the PCM is “prepared”. You can set the format type, sample rate, etc. here. The difference from ``hw_params`` is that the ``prepare`` callback will be called each time :c:func:`snd_pcm_prepare()` is called, i.e. when recovering after underruns, etc. -Note that this callback is now non-atomic. You can use +Note that this callback is non-atomic. You can use schedule-related functions safely in this callback. In this and the following callbacks, you can refer to the values via @@ -2026,13 +1932,11 @@ trigger callback static int snd_xxx_trigger(struct snd_pcm_substream *substream, int cmd); -This is called when the pcm is started, stopped or paused. - -Which action is specified in the second argument, -``SNDRV_PCM_TRIGGER_XXX`` in ``<sound/pcm.h>``. At least, the ``START`` -and ``STOP`` commands must be defined in this callback. +This is called when the PCM is started, stopped or paused. -:: +The action is specified in the second argument, ``SNDRV_PCM_TRIGGER_XXX`` +defined in ``<sound/pcm.h>``. At least, the ``START`` +and ``STOP`` commands must be defined in this callback:: switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -2045,23 +1949,23 @@ and ``STOP`` commands must be defined in this callback. return -EINVAL; } -When the pcm supports the pause operation (given in the info field of +When the PCM supports the pause operation (given in the info field of the hardware table), the ``PAUSE_PUSH`` and ``PAUSE_RELEASE`` commands -must be handled here, too. The former is the command to pause the pcm, -and the latter to restart the pcm again. +must be handled here, too. The former is the command to pause the PCM, +and the latter to restart the PCM again. -When the pcm supports the suspend/resume operation, regardless of full +When the PCM supports the suspend/resume operation, regardless of full or partial suspend/resume support, the ``SUSPEND`` and ``RESUME`` commands must be handled, too. These commands are issued when the power-management status is changed. Obviously, the ``SUSPEND`` and -``RESUME`` commands suspend and resume the pcm substream, and usually, +``RESUME`` commands suspend and resume the PCM substream, and usually, they are identical to the ``STOP`` and ``START`` commands, respectively. See the `Power Management`_ section for details. -As mentioned, this callback is atomic as default unless ``nonatomic`` +As mentioned, this callback is atomic by default unless the ``nonatomic`` flag set, and you cannot call functions which may sleep. The ``trigger`` callback should be as minimal as possible, just really -triggering the DMA. The other stuff should be initialized +triggering the DMA. The other stuff should be initialized in ``hw_params`` and ``prepare`` callbacks properly beforehand. sync_stop callback @@ -2072,22 +1976,22 @@ sync_stop callback static int snd_xxx_sync_stop(struct snd_pcm_substream *substream); This callback is optional, and NULL can be passed. It's called after -the PCM core stops the stream and changes the stream state +the PCM core stops the stream, before it changes the stream state via ``prepare``, ``hw_params`` or ``hw_free``. Since the IRQ handler might be still pending, we need to wait until the pending task finishes before moving to the next step; otherwise it -might lead to a crash due to resource conflicts or access to the freed +might lead to a crash due to resource conflicts or access to freed resources. A typical behavior is to call a synchronization function like :c:func:`synchronize_irq()` here. -For majority of drivers that need only a call of +For the majority of drivers that need only a call of :c:func:`synchronize_irq()`, there is a simpler setup, too. -While keeping NULL to ``sync_stop`` PCM callback, the driver can set -``card->sync_irq`` field to store the valid interrupt number after -requesting an IRQ, instead. Then PCM core will look call +While keeping the ``sync_stop`` PCM callback NULL, the driver can set +the ``card->sync_irq`` field to the returned interrupt number after +requesting an IRQ, instead. Then PCM core will call :c:func:`synchronize_irq()` with the given IRQ appropriately. -If the IRQ handler is released at the card destructor, you don't need +If the IRQ handler is released by the card destructor, you don't need to clear ``card->sync_irq``, as the card itself is being released. So, usually you'll need to add just a single line for assigning ``card->sync_irq`` in the driver code unless the driver re-acquires @@ -2103,30 +2007,30 @@ pointer callback static snd_pcm_uframes_t snd_xxx_pointer(struct snd_pcm_substream *substream) This callback is called when the PCM middle layer inquires the current -hardware position on the buffer. The position must be returned in +hardware position in the buffer. The position must be returned in frames, ranging from 0 to ``buffer_size - 1``. -This is called usually from the buffer-update routine in the pcm +This is usually called from the buffer-update routine in the PCM middle layer, which is invoked when :c:func:`snd_pcm_period_elapsed()` -is called in the interrupt routine. Then the pcm middle layer updates +is called by the interrupt routine. Then the PCM middle layer updates the position and calculates the available space, and wakes up the sleeping poll threads, etc. -This callback is also atomic as default. +This callback is also atomic by default. copy_user, copy_kernel and fill_silence ops ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ These callbacks are not mandatory, and can be omitted in most cases. These callbacks are used when the hardware buffer cannot be in the -normal memory space. Some chips have their own buffer on the hardware +normal memory space. Some chips have their own buffer in the hardware which is not mappable. In such a case, you have to transfer the data manually from the memory buffer to the hardware buffer. Or, if the buffer is non-contiguous on both physical and virtual memory spaces, these callbacks must be defined, too. If these two callbacks are defined, copy and set-silence operations -are done by them. The detailed will be described in the later section +are done by them. The details will be described in the later section `Buffer and Memory Management`_. ack callback @@ -2137,7 +2041,11 @@ This callback is also not mandatory. This callback is called when the emu10k1-fx and cs46xx need to track the current ``appl_ptr`` for the internal buffer, and this callback is useful only for such a purpose. -This callback is atomic as default. +The callback function may return 0 or a negative error. When the +return value is ``-EPIPE``, PCM core treats that as a buffer XRUN, +and changes the state to ``SNDRV_PCM_STATE_XRUN`` automatically. + +This callback is atomic by default. page callback ~~~~~~~~~~~~~ @@ -2145,16 +2053,15 @@ page callback This callback is optional too. The mmap calls this callback to get the page fault address. -Since the recent changes, you need no special callback any longer for -the standard SG-buffer or vmalloc-buffer. Hence this callback should -be rarely used. +You need no special callback for the standard SG-buffer or vmalloc- +buffer. Hence this callback should be rarely used. -mmap calllback -~~~~~~~~~~~~~~ +mmap callback +~~~~~~~~~~~~~ This is another optional callback for controlling mmap behavior. -Once when defined, PCM core calls this callback when a page is -memory-mapped instead of dealing via the standard helper. +When defined, the PCM core calls this callback when a page is +memory-mapped, instead of using the standard helper. If you need special handling (due to some architecture or device-specific issues), implement everything here as you like. @@ -2162,13 +2069,14 @@ device-specific issues), implement everything here as you like. PCM Interrupt Handler --------------------- -The rest of pcm stuff is the PCM interrupt handler. The role of PCM +The remainder of the PCM stuff is the PCM interrupt handler. The role +of the PCM interrupt handler in the sound driver is to update the buffer position and to tell the PCM middle layer when the buffer position goes across -the prescribed period size. To inform this, call the +the specified period boundary. To inform about this, call the :c:func:`snd_pcm_period_elapsed()` function. -There are several types of sound chips to generate the interrupts. +There are several ways sound chips can generate interrupts. Interrupts at the period (fragment) boundary ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ @@ -2184,14 +2092,12 @@ chip record to hold the current running substream pointer, and set the pointer value at ``open`` callback (and reset at ``close`` callback). If you acquire a spinlock in the interrupt handler, and the lock is used -in other pcm callbacks, too, then you have to release the lock before +in other PCM callbacks, too, then you have to release the lock before calling :c:func:`snd_pcm_period_elapsed()`, because -:c:func:`snd_pcm_period_elapsed()` calls other pcm callbacks +:c:func:`snd_pcm_period_elapsed()` calls other PCM callbacks inside. -Typical code would be like: - -:: +Typical code would look like:: static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id) @@ -2211,6 +2117,12 @@ Typical code would be like: return IRQ_HANDLED; } +Also, when the device can detect a buffer underrun/overrun, the driver +can notify the XRUN status to the PCM core by calling +:c:func:`snd_pcm_stop_xrun()`. This function stops the stream and sets +the PCM state to ``SNDRV_PCM_STATE_XRUN``. Note that it must be called +outside the PCM stream lock, hence it can't be called from the atomic +callback. High frequency timer interrupts @@ -2223,9 +2135,7 @@ position and accumulate the processed sample length at each interrupt. When the accumulated size exceeds the period size, call :c:func:`snd_pcm_period_elapsed()` and reset the accumulator. -Typical code would be like the following. - -:: +Typical code would look as follows:: static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id) @@ -2270,9 +2180,9 @@ Typical code would be like the following. On calling :c:func:`snd_pcm_period_elapsed()` ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ -In both cases, even if more than one period are elapsed, you don't have +In both cases, even if more than one period has elapsed, you don't have to call :c:func:`snd_pcm_period_elapsed()` many times. Call only -once. And the pcm layer will check the current hardware pointer and +once. And the PCM layer will check the current hardware pointer and update to the latest status. Atomicity @@ -2283,15 +2193,16 @@ kernel programming are race conditions. In the Linux kernel, they are usually avoided via spin-locks, mutexes or semaphores. In general, if a race condition can happen in an interrupt handler, it has to be managed atomically, and you have to use a spinlock to protect the critical -session. If the critical section is not in interrupt handler code and if +section. If the critical section is not in interrupt handler code and if taking a relatively long time to execute is acceptable, you should use mutexes or semaphores instead. -As already seen, some pcm callbacks are atomic and some are not. For -example, the ``hw_params`` callback is non-atomic, while ``trigger`` +As already seen, some PCM callbacks are atomic and some are not. For +example, the ``hw_params`` callback is non-atomic, while the ``trigger`` callback is atomic. This means, the latter is called already in a -spinlock held by the PCM middle layer. Please take this atomicity into -account when you choose a locking scheme in the callbacks. +spinlock held by the PCM middle layer, the PCM stream lock. Please +take this atomicity into account when you choose a locking scheme in +the callbacks. In the atomic callbacks, you cannot use functions which may call :c:func:`schedule()` or go to :c:func:`sleep()`. Semaphores and @@ -2302,29 +2213,34 @@ callback, please use :c:func:`udelay()` or :c:func:`mdelay()`. All three atomic callbacks (trigger, pointer, and ack) are called with local interrupts disabled. -The recent changes in PCM core code, however, allow all PCM operations -to be non-atomic. This assumes that the all caller sides are in +However, it is possible to request all PCM operations to be non-atomic. +This assumes that all call sites are in non-atomic contexts. For example, the function :c:func:`snd_pcm_period_elapsed()` is called typically from the interrupt handler. But, if you set up the driver to use a threaded interrupt handler, this call can be in non-atomic context, too. In such -a case, you can set ``nonatomic`` filed of struct snd_pcm object +a case, you can set the ``nonatomic`` field of the struct snd_pcm object after creating it. When this flag is set, mutex and rwsem are used internally in the PCM core instead of spin and rwlocks, so that you can call all PCM functions safely in a non-atomic context. +Also, in some cases, you might need to call +:c:func:`snd_pcm_period_elapsed()` in the atomic context (e.g. the +period gets elapsed during ``ack`` or other callback). There is a +variant that can be called inside the PCM stream lock +:c:func:`snd_pcm_period_elapsed_under_stream_lock()` for that purpose, +too. + Constraints ----------- -If your chip supports unconventional sample rates, or only the limited -samples, you need to set a constraint for the condition. +Due to physical limitations, hardware is not infinitely configurable. +These limitations are expressed by setting constraints. -For example, in order to restrict the sample rates in the some supported +For example, in order to restrict the sample rates to some supported values, use :c:func:`snd_pcm_hw_constraint_list()`. You need to -call this function in the open callback. - -:: +call this function in the open callback:: static unsigned int rates[] = {4000, 10000, 22050, 44100}; @@ -2346,16 +2262,12 @@ call this function in the open callback. .... } - - There are many different constraints. Look at ``sound/pcm.h`` for a complete list. You can even define your own constraint rules. For example, let's suppose my_chip can manage a substream of 1 channel if and only if the format is ``S16_LE``, otherwise it supports any format -specified in struct snd_pcm_hardware> (or in any other -constraint_list). You can build a rule like this: - -:: +specified in struct snd_pcm_hardware (or in any other +constraint_list). You can build a rule like this:: static int hw_rule_channels_by_format(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) @@ -2375,9 +2287,7 @@ constraint_list). You can build a rule like this: } -Then you need to call this function to add your rule: - -:: +Then you need to call this function to add your rule:: snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, hw_rule_channels_by_format, NULL, @@ -2386,9 +2296,7 @@ Then you need to call this function to add your rule: The rule function is called when an application sets the PCM format, and it refines the number of channels accordingly. But an application may set the number of channels before setting the format. Thus you also need -to define the inverse rule: - -:: +to define the inverse rule:: static int hw_rule_format_by_channels(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) @@ -2407,16 +2315,14 @@ to define the inverse rule: } -... and in the open callback: - -:: +... and in the open callback:: snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, hw_rule_format_by_channels, NULL, SNDRV_PCM_HW_PARAM_CHANNELS, -1); One typical usage of the hw constraints is to align the buffer size -with the period size. As default, ALSA PCM core doesn't enforce the +with the period size. By default, ALSA PCM core doesn't enforce the buffer size to be aligned with the period size. For example, it'd be possible to have a combination like 256 period bytes with 999 buffer bytes. @@ -2424,9 +2330,7 @@ bytes. Many device chips, however, require the buffer to be a multiple of periods. In such a case, call :c:func:`snd_pcm_hw_constraint_integer()` for -``SNDRV_PCM_HW_PARAM_PERIODS``. - -:: +``SNDRV_PCM_HW_PARAM_PERIODS``:: snd_pcm_hw_constraint_integer(substream->runtime, SNDRV_PCM_HW_PARAM_PERIODS); @@ -2434,7 +2338,7 @@ periods. In such a case, call This assures that the number of periods is integer, hence the buffer size is aligned with the period size. -The hw constraint is a very much powerful mechanism to define the +The hw constraint is a very powerful mechanism to define the preferred PCM configuration, and there are relevant helpers. I won't give more details here, rather I would like to say, “Luke, use the source.” @@ -2461,9 +2365,7 @@ Definition of Controls To create a new control, you need to define the following three callbacks: ``info``, ``get`` and ``put``. Then, define a -struct snd_kcontrol_new record, such as: - -:: +struct snd_kcontrol_new record, such as:: static struct snd_kcontrol_new my_control = { @@ -2506,7 +2408,7 @@ The ``private_value`` field contains an arbitrary long integer value for this record. When using the generic ``info``, ``get`` and ``put`` callbacks, you can pass a value through this field. If several small numbers are necessary, you can combine them in bitwise. Or, it's -possible to give a pointer (casted to unsigned long) of some record to +possible to store a pointer (casted to unsigned long) of some record in this field, too. The ``tlv`` field can be used to provide metadata about the control; @@ -2573,7 +2475,7 @@ The access flag is the bitmask which specifies the access type of the given control. The default access type is ``SNDRV_CTL_ELEM_ACCESS_READWRITE``, which means both read and write are allowed to this control. When the access flag is omitted (i.e. = 0), it -is considered as ``READWRITE`` access as default. +is considered as ``READWRITE`` access by default. When the control is read-only, pass ``SNDRV_CTL_ELEM_ACCESS_READ`` instead. In this case, you don't have to define the ``put`` callback. @@ -2586,8 +2488,11 @@ If the control value changes frequently (e.g. the VU meter), changed without `Change notification`_. Applications should poll such a control constantly. -When the control is inactive, set the ``INACTIVE`` flag, too. There are -``LOCK`` and ``OWNER`` flags to change the write permissions. +When the control may be updated, but currently has no effect on anything, +setting the ``INACTIVE`` flag may be appropriate. For example, PCM +controls should be inactive while no PCM device is open. + +There are ``LOCK`` and ``OWNER`` flags to change the write permissions. Control Callbacks ----------------- @@ -2598,9 +2503,7 @@ info callback The ``info`` callback is used to get detailed information on this control. This must store the values of the given struct snd_ctl_elem_info object. For example, -for a boolean control with a single element: - -:: +for a boolean control with a single element:: static int snd_myctl_mono_info(struct snd_kcontrol *kcontrol, @@ -2619,13 +2522,11 @@ The ``type`` field specifies the type of the control. There are ``BOOLEAN``, ``INTEGER``, ``ENUMERATED``, ``BYTES``, ``IEC958`` and ``INTEGER64``. The ``count`` field specifies the number of elements in this control. For example, a stereo volume would have count = 2. The -``value`` field is a union, and the values stored are depending on the +``value`` field is a union, and the values stored depend on the type. The boolean and integer types are identical. -The enumerated type is a bit different from others. You'll need to set -the string for the currently given item index. - -:: +The enumerated type is a bit different from the others. You'll need to +set the string for the selectec item index:: static int snd_myctl_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) @@ -2670,13 +2571,10 @@ stereo channel boolean item. get callback ~~~~~~~~~~~~ -This callback is used to read the current value of the control and to -return to user-space. - -For example, - -:: +This callback is used to read the current value of the control, so it +can be returned to user-space. +For example:: static int snd_myctl_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -2691,15 +2589,11 @@ For example, The ``value`` field depends on the type of control as well as on the info callback. For example, the sb driver uses this field to store the register offset, the bit-shift and the bit-mask. The ``private_value`` -field is set as follows: - -:: +field is set as follows:: .private_value = reg | (shift << 16) | (mask << 24) -and is retrieved in callbacks like - -:: +and is retrieved in callbacks like:: static int snd_sbmixer_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -2711,19 +2605,16 @@ and is retrieved in callbacks like } In the ``get`` callback, you have to fill all the elements if the -control has more than one elements, i.e. ``count > 1``. In the example +control has more than one element, i.e. ``count > 1``. In the example above, we filled only one element (``value.integer.value[0]``) since -it's assumed as ``count = 1``. +``count = 1`` is assumed. put callback ~~~~~~~~~~~~ -This callback is used to write a value from user-space. - -For example, - -:: +This callback is used to write a value coming from user-space. +For example:: static int snd_myctl_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -2746,12 +2637,12 @@ value is not changed, return 0 instead. If any fatal error happens, return a negative error code as usual. As in the ``get`` callback, when the control has more than one -elements, all elements must be evaluated in this callback, too. +element, all elements must be evaluated in this callback, too. Callbacks are not atomic ~~~~~~~~~~~~~~~~~~~~~~~~ -All these three callbacks are basically not atomic. +All these three callbacks are not-atomic. Control Constructor ------------------- @@ -2760,9 +2651,7 @@ When everything is ready, finally we can create a new control. To create a control, there are two functions to be called, :c:func:`snd_ctl_new1()` and :c:func:`snd_ctl_add()`. -In the simplest way, you can do like this: - -:: +In the simplest way, you can do it like this:: err = snd_ctl_add(card, snd_ctl_new1(&my_control, chip)); if (err < 0) @@ -2780,9 +2669,7 @@ Change Notification ------------------- If you need to change and update a control in the interrupt routine, you -can call :c:func:`snd_ctl_notify()`. For example, - -:: +can call :c:func:`snd_ctl_notify()`. For example:: snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, id_pointer); @@ -2796,13 +2683,11 @@ for hardware volume interrupts. Metadata -------- -To provide information about the dB values of a mixer control, use on of +To provide information about the dB values of a mixer control, use one of the ``DECLARE_TLV_xxx`` macros from ``<sound/tlv.h>`` to define a variable containing this information, set the ``tlv.p`` field to point to this variable, and include the ``SNDRV_CTL_ELEM_ACCESS_TLV_READ`` flag -in the ``access`` field; like this: - -:: +in the ``access`` field; like this:: static DECLARE_TLV_DB_SCALE(db_scale_my_control, -4050, 150, 0); @@ -2892,9 +2777,7 @@ AC97 Constructor ---------------- To create an ac97 instance, first call :c:func:`snd_ac97_bus()` -with an ``ac97_bus_ops_t`` record with callback functions. - -:: +with an ``ac97_bus_ops_t`` record with callback functions:: struct snd_ac97_bus *bus; static struct snd_ac97_bus_ops ops = { @@ -2906,10 +2789,8 @@ with an ``ac97_bus_ops_t`` record with callback functions. The bus record is shared among all belonging ac97 instances. -And then call :c:func:`snd_ac97_mixer()` with an struct snd_ac97_template -record together with the bus pointer created above. - -:: +And then call :c:func:`snd_ac97_mixer()` with a struct snd_ac97_template +record together with the bus pointer created above:: struct snd_ac97_template ac97; int err; @@ -2934,9 +2815,7 @@ correspond to the functions for read and write accesses to the hardware low-level codes. The ``read`` callback returns the register value specified in the -argument. - -:: +argument:: static unsigned short snd_mychip_ac97_read(struct snd_ac97 *ac97, unsigned short reg) @@ -2949,9 +2828,7 @@ argument. Here, the chip can be cast from ``ac97->private_data``. Meanwhile, the ``write`` callback is used to set the register -value - -:: +value:: static void snd_mychip_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) @@ -2984,32 +2861,24 @@ Both :c:func:`snd_ac97_write()` and the given register (``AC97_XXX``). The difference between them is that :c:func:`snd_ac97_update()` doesn't write a value if the given value has been already set, while :c:func:`snd_ac97_write()` -always rewrites the value. - -:: +always rewrites the value:: snd_ac97_write(ac97, AC97_MASTER, 0x8080); snd_ac97_update(ac97, AC97_MASTER, 0x8080); :c:func:`snd_ac97_read()` is used to read the value of the given -register. For example, - -:: +register. For example:: value = snd_ac97_read(ac97, AC97_MASTER); :c:func:`snd_ac97_update_bits()` is used to update some bits in -the given register. - -:: +the given register:: snd_ac97_update_bits(ac97, reg, mask, value); Also, there is a function to change the sample rate (of a given register such as ``AC97_PCM_FRONT_DAC_RATE``) when VRA or DRA is supported by the -codec: :c:func:`snd_ac97_set_rate()`. - -:: +codec: :c:func:`snd_ac97_set_rate()`:: snd_ac97_set_rate(ac97, AC97_PCM_FRONT_DAC_RATE, 44100); @@ -3064,9 +2933,7 @@ mpu401 stuff. For example, emu10k1 has its own mpu401 routines. MIDI Constructor ---------------- -To create a rawmidi object, call :c:func:`snd_mpu401_uart_new()`. - -:: +To create a rawmidi object, call :c:func:`snd_mpu401_uart_new()`:: struct snd_rawmidi *rmidi; snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, port, info_flags, @@ -3111,16 +2978,12 @@ corresponds to the data port. If not, you may change the ``cport`` field of struct snd_mpu401 manually afterward. However, struct snd_mpu401 pointer is not returned explicitly by :c:func:`snd_mpu401_uart_new()`. You -need to cast ``rmidi->private_data`` to struct snd_mpu401 explicitly, - -:: +need to cast ``rmidi->private_data`` to struct snd_mpu401 explicitly:: struct snd_mpu401 *mpu; mpu = rmidi->private_data; -and reset the ``cport`` as you like: - -:: +and reset the ``cport`` as you like:: mpu->cport = my_own_control_port; @@ -3144,9 +3007,7 @@ occurred. In this case, you need to pass the private_data of the returned rawmidi object from :c:func:`snd_mpu401_uart_new()` as the second -argument of :c:func:`snd_mpu401_uart_interrupt()`. - -:: +argument of :c:func:`snd_mpu401_uart_interrupt()`:: snd_mpu401_uart_interrupt(irq, rmidi->private_data, regs); @@ -3170,9 +3031,7 @@ RawMIDI Constructor ------------------- To create a rawmidi device, call the :c:func:`snd_rawmidi_new()` -function: - -:: +function:: struct snd_rawmidi *rmidi; err = snd_rawmidi_new(chip->card, "MyMIDI", 0, outs, ins, &rmidi); @@ -3202,16 +3061,12 @@ output and input at the same time. After the rawmidi device is created, you need to set the operators (callbacks) for each substream. There are helper functions to set the -operators for all the substreams of a device: - -:: +operators for all the substreams of a device:: snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &snd_mymidi_output_ops); snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &snd_mymidi_input_ops); -The operators are usually defined like this: - -:: +The operators are usually defined like this:: static struct snd_rawmidi_ops snd_mymidi_output_ops = { .open = snd_mymidi_output_open, @@ -3222,9 +3077,7 @@ The operators are usually defined like this: These callbacks are explained in the `RawMIDI Callbacks`_ section. If there are more than one substream, you should give a unique name to -each of them: - -:: +each of them:: struct snd_rawmidi_substream *substream; list_for_each_entry(substream, @@ -3242,9 +3095,7 @@ device can be accessed as ``substream->rmidi->private_data``. If there is more than one port, your callbacks can determine the port index from the struct snd_rawmidi_substream data passed to each -callback: - -:: +callback:: struct snd_rawmidi_substream *substream; int index = substream->number; @@ -3289,9 +3140,7 @@ of bytes that have been read; this will be less than the number of bytes requested when there are no more data in the buffer. After the data have been transmitted successfully, call :c:func:`snd_rawmidi_transmit_ack()` to remove the data from the -substream buffer: - -:: +substream buffer:: unsigned char data; while (snd_rawmidi_transmit_peek(substream, &data, 1) == 1) { @@ -3303,9 +3152,7 @@ substream buffer: If you know beforehand that the hardware will accept data, you can use the :c:func:`snd_rawmidi_transmit()` function which reads some -data and removes them from the buffer at once: - -:: +data and removes them from the buffer at once:: while (snd_mychip_transmit_possible()) { unsigned char data; @@ -3340,9 +3187,7 @@ The ``trigger`` callback must not sleep; the actual reading of data from the device is usually done in an interrupt handler. When data reception is enabled, your interrupt handler should call -:c:func:`snd_rawmidi_receive()` for all received data: - -:: +:c:func:`snd_rawmidi_receive()` for all received data:: void snd_mychip_midi_interrupt(...) { @@ -3388,9 +3233,7 @@ whereas in OSS compatible mode, FM registers can be accessed with the OSS direct-FM compatible API in ``/dev/dmfmX`` device. To create the OPL3 component, you have two functions to call. The first -one is a constructor for the ``opl3_t`` instance. - -:: +one is a constructor for the ``opl3_t`` instance:: struct snd_opl3 *opl3; snd_opl3_create(card, lport, rport, OPL3_HW_OPL3_XXX, @@ -3408,9 +3251,7 @@ the opl3 module will allocate the specified ports by itself. When the accessing the hardware requires special method instead of the standard I/O access, you can create opl3 instance separately with -:c:func:`snd_opl3_new()`. - -:: +:c:func:`snd_opl3_new()`:: struct snd_opl3 *opl3; snd_opl3_new(card, OPL3_HW_OPL3_XXX, &opl3); @@ -3427,9 +3268,7 @@ proper state. Note that :c:func:`snd_opl3_create()` always calls it internally. If the opl3 instance is created successfully, then create a hwdep device -for this opl3. - -:: +for this opl3:: struct snd_hwdep *opl3hwdep; snd_opl3_hwdep_new(opl3, 0, 1, &opl3hwdep); @@ -3451,9 +3290,7 @@ the micro code. In such a case, you can create a hwdep ``isa/sb/sb16_csp.c``. The creation of the ``hwdep`` instance is done via -:c:func:`snd_hwdep_new()`. - -:: +:c:func:`snd_hwdep_new()`:: struct snd_hwdep *hw; snd_hwdep_new(card, "My HWDEP", 0, &hw); @@ -3461,18 +3298,14 @@ The creation of the ``hwdep`` instance is done via where the third argument is the index number. You can then pass any pointer value to the ``private_data``. If you -assign a private data, you should define the destructor, too. The -destructor function is set in the ``private_free`` field. - -:: +assign private data, you should define a destructor, too. The +destructor function is set in the ``private_free`` field:: struct mydata *p = kmalloc(sizeof(*p), GFP_KERNEL); hw->private_data = p; hw->private_free = mydata_free; -and the implementation of the destructor would be: - -:: +and the implementation of the destructor would be:: static void mydata_free(struct snd_hwdep *hw) { @@ -3482,9 +3315,7 @@ and the implementation of the destructor would be: The arbitrary file operations can be defined for this instance. The file operators are defined in the ``ops`` table. For example, assume that -this chip needs an ioctl. - -:: +this chip needs an ioctl:: hw->ops.open = mydata_open; hw->ops.ioctl = mydata_ioctl; @@ -3534,31 +3365,30 @@ Buffer Types ALSA provides several different buffer allocation functions depending on the bus and the architecture. All these have a consistent API. The -allocation of physically-contiguous pages is done via +allocation of physically-contiguous pages is done via the :c:func:`snd_malloc_xxx_pages()` function, where xxx is the bus type. -The allocation of pages with fallback is -:c:func:`snd_malloc_xxx_pages_fallback()`. This function tries -to allocate the specified pages but if the pages are not available, it -tries to reduce the page sizes until enough space is found. +The allocation of pages with fallback is done via +:c:func:`snd_dma_alloc_pages_fallback()`. This function tries +to allocate the specified number of pages, but if not enough pages are +available, it tries to reduce the request size until enough space +is found, down to one page. -The release the pages, call :c:func:`snd_free_xxx_pages()` +To release the pages, call the :c:func:`snd_dma_free_pages()` function. Usually, ALSA drivers try to allocate and reserve a large contiguous -physical space at the time the module is loaded for the later use. This +physical space at the time the module is loaded for later use. This is called “pre-allocation”. As already written, you can call the -following function at pcm instance construction time (in the case of PCI -bus). - -:: +following function at PCM instance construction time (in the case of PCI +bus):: snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, &pci->dev, size, max); -where ``size`` is the byte size to be pre-allocated and the ``max`` is -the maximum size to be changed via the ``prealloc`` proc file. The +where ``size`` is the byte size to be pre-allocated and ``max`` is +the maximum size settable via the ``prealloc`` proc file. The allocator will try to get an area as large as possible within the given size. @@ -3567,10 +3397,10 @@ dependent on the bus. For normal devices, pass the device pointer (typically identical as ``card->dev``) to the third argument with ``SNDRV_DMA_TYPE_DEV`` type. -For the continuous buffer unrelated to the +A continuous buffer unrelated to the bus can be pre-allocated with ``SNDRV_DMA_TYPE_CONTINUOUS`` type. You can pass NULL to the device pointer in that case, which is the -default mode implying to allocate with ``GFP_KERNEL`` flag. +default mode implying to allocate with the ``GFP_KERNEL`` flag. If you need a restricted (lower) address, set up the coherent DMA mask bits for the device, and pass the device pointer, like the normal device memory allocations. For this type, it's still allowed to pass @@ -3580,37 +3410,33 @@ For the scatter-gather buffers, use ``SNDRV_DMA_TYPE_DEV_SG`` with the device pointer (see the `Non-Contiguous Buffers`_ section). Once the buffer is pre-allocated, you can use the allocator in the -``hw_params`` callback: - -:: +``hw_params`` callback:: snd_pcm_lib_malloc_pages(substream, size); Note that you have to pre-allocate to use this function. -Most of drivers use, though, rather the newly introduced "managed -buffer allocation mode" instead of the manual allocation or release. +But most drivers use the "managed buffer allocation mode" instead +of manual allocation and release. This is done by calling :c:func:`snd_pcm_set_managed_buffer_all()` -instead of :c:func:`snd_pcm_lib_preallocate_pages_for_all()`. - -:: +instead of :c:func:`snd_pcm_lib_preallocate_pages_for_all()`:: snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV, &pci->dev, size, max); -where passed arguments are identical in both functions. +where the passed arguments are identical for both functions. The difference in the managed mode is that PCM core will call :c:func:`snd_pcm_lib_malloc_pages()` internally already before calling the PCM ``hw_params`` callback, and call :c:func:`snd_pcm_lib_free_pages()` after the PCM ``hw_free`` callback automatically. So the driver doesn't have to call these functions explicitly in its callback any -longer. This made many driver code having NULL ``hw_params`` and +longer. This allows many drivers to have NULL ``hw_params`` and ``hw_free`` entries. External Hardware Buffers ------------------------- -Some chips have their own hardware buffers and the DMA transfer from the +Some chips have their own hardware buffers and DMA transfer from the host memory is not available. In such a case, you need to either 1) copy/set the audio data directly to the external hardware buffer, or 2) make an intermediate buffer and copy/set the data from it to the @@ -3618,8 +3444,8 @@ external hardware buffer in interrupts (or in tasklets, preferably). The first case works fine if the external hardware buffer is large enough. This method doesn't need any extra buffers and thus is more -effective. You need to define the ``copy_user`` and ``copy_kernel`` -callbacks for the data transfer, in addition to ``fill_silence`` +efficient. You need to define the ``copy_user`` and ``copy_kernel`` +callbacks for the data transfer, in addition to the ``fill_silence`` callback for playback. However, there is a drawback: it cannot be mmapped. The examples are GUS's GF1 PCM or emu8000's wavetable PCM. @@ -3633,16 +3459,14 @@ buffer instead of the host memory. In this case, mmap is available only on certain architectures like the Intel one. In non-mmap mode, the data cannot be transferred as in the normal way. Thus you need to define the ``copy_user``, ``copy_kernel`` and ``fill_silence`` callbacks as well, -as in the cases above. The examples are found in ``rme32.c`` and +as in the cases above. Examples are found in ``rme32.c`` and ``rme96.c``. The implementation of the ``copy_user``, ``copy_kernel`` and ``silence`` callbacks depends upon whether the hardware supports interleaved or non-interleaved samples. The ``copy_user`` callback is -defined like below, a bit differently depending whether the direction -is playback or capture: - -:: +defined like below, a bit differently depending on whether the direction +is playback or capture:: static int playback_copy_user(struct snd_pcm_substream *substream, int channel, unsigned long pos, @@ -3652,8 +3476,7 @@ is playback or capture: void __user *dst, unsigned long count); In the case of interleaved samples, the second argument (``channel``) is -not used. The third argument (``pos``) points the current position -offset in bytes. +not used. The third argument (``pos``) specifies the position in bytes. The meaning of the fourth argument is different between playback and capture. For playback, it holds the source data pointer, and for @@ -3664,49 +3487,42 @@ The last argument is the number of bytes to be copied. What you have to do in this callback is again different between playback and capture directions. In the playback case, you copy the given amount of data (``count``) at the specified pointer (``src``) to the specified -offset (``pos``) on the hardware buffer. When coded like memcpy-like -way, the copy would be like: - -:: +offset (``pos``) in the hardware buffer. When coded like memcpy-like +way, the copy would look like:: my_memcpy_from_user(my_buffer + pos, src, count); For the capture direction, you copy the given amount of data (``count``) -at the specified offset (``pos``) on the hardware buffer to the -specified pointer (``dst``). - -:: +at the specified offset (``pos``) in the hardware buffer to the +specified pointer (``dst``):: my_memcpy_to_user(dst, my_buffer + pos, count); -Here the functions are named as ``from_user`` and ``to_user`` because +Here the functions are named ``from_user`` and ``to_user`` because it's the user-space buffer that is passed to these callbacks. That -is, the callback is supposed to copy from/to the user-space data +is, the callback is supposed to copy data from/to the user-space directly to/from the hardware buffer. Careful readers might notice that these callbacks receive the arguments in bytes, not in frames like other callbacks. It's because -it would make coding easier like the examples above, and also it makes -easier to unify both the interleaved and non-interleaved cases, as -explained in the following. +this makes coding easier like in the examples above, and also it makes +it easier to unify both the interleaved and non-interleaved cases, as +explained below. In the case of non-interleaved samples, the implementation will be a bit -more complicated. The callback is called for each channel, passed by -the second argument, so totally it's called for N-channels times per -transfer. - -The meaning of other arguments are almost same as the interleaved -case. The callback is supposed to copy the data from/to the given -user-space buffer, but only for the given channel. For the detailed -implementations, please check ``isa/gus/gus_pcm.c`` or -"pci/rme9652/rme9652.c" as examples. +more complicated. The callback is called for each channel, passed in +the second argument, so in total it's called N times per transfer. -The above callbacks are the copy from/to the user-space buffer. There -are some cases where we want copy from/to the kernel-space buffer -instead. In such a case, ``copy_kernel`` callback is called. It'd -look like: +The meaning of the other arguments are almost the same as in the +interleaved case. The callback is supposed to copy the data from/to +the given user-space buffer, but only for the given channel. For +details, please check ``isa/gus/gus_pcm.c`` or ``pci/rme9652/rme9652.c`` +as examples. -:: +The above callbacks are the copies from/to the user-space buffer. There +are some cases where we want to copy from/to the kernel-space buffer +instead. In such a case, the ``copy_kernel`` callback is called. It'd +look like:: static int playback_copy_kernel(struct snd_pcm_substream *substream, int channel, unsigned long pos, @@ -3716,19 +3532,15 @@ look like: void *dst, unsigned long count); As found easily, the only difference is that the buffer pointer is -without ``__user`` prefix; that is, a kernel-buffer pointer is passed +without a ``__user`` prefix; that is, a kernel-buffer pointer is passed in the fourth argument. Correspondingly, the implementation would be -a version without the user-copy, such as: - -:: +a version without the user-copy, such as:: my_memcpy(my_buffer + pos, src, count); Usually for the playback, another callback ``fill_silence`` is defined. It's implemented in a similar way as the copy callbacks -above: - -:: +above:: static int silence(struct snd_pcm_substream *substream, int channel, unsigned long pos, unsigned long count); @@ -3736,54 +3548,47 @@ above: The meanings of arguments are the same as in the ``copy_user`` and ``copy_kernel`` callbacks, although there is no buffer pointer argument. In the case of interleaved samples, the channel argument has -no meaning, as well as on ``copy_*`` callbacks. +no meaning, as for the ``copy_*`` callbacks. -The role of ``fill_silence`` callback is to set the given amount -(``count``) of silence data at the specified offset (``pos``) on the +The role of the ``fill_silence`` callback is to set the given amount +(``count``) of silence data at the specified offset (``pos``) in the hardware buffer. Suppose that the data format is signed (that is, the silent-data is 0), and the implementation using a memset-like function -would be like: - -:: +would look like:: my_memset(my_buffer + pos, 0, count); In the case of non-interleaved samples, again, the implementation -becomes a bit more complicated, as it's called N-times per transfer +becomes a bit more complicated, as it's called N times per transfer for each channel. See, for example, ``isa/gus/gus_pcm.c``. Non-Contiguous Buffers ---------------------- -If your hardware supports the page table as in emu10k1 or the buffer -descriptors as in via82xx, you can use the scatter-gather (SG) DMA. ALSA +If your hardware supports a page table as in emu10k1 or buffer +descriptors as in via82xx, you can use scatter-gather (SG) DMA. ALSA provides an interface for handling SG-buffers. The API is provided in ``<sound/pcm.h>``. For creating the SG-buffer handler, call :c:func:`snd_pcm_set_managed_buffer()` or :c:func:`snd_pcm_set_managed_buffer_all()` with -``SNDRV_DMA_TYPE_DEV_SG`` in the PCM constructor like other PCI -pre-allocator. You need to pass ``&pci->dev``, where pci is -the struct pci_dev pointer of the chip as -well. - -:: +``SNDRV_DMA_TYPE_DEV_SG`` in the PCM constructor like for other PCI +pre-allocations. You need to pass ``&pci->dev``, where pci is +the struct pci_dev pointer of the chip as well:: snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV_SG, &pci->dev, size, max); The ``struct snd_sg_buf`` instance is created as -``substream->dma_private`` in turn. You can cast the pointer like: - -:: +``substream->dma_private`` in turn. You can cast the pointer like:: struct snd_sg_buf *sgbuf = (struct snd_sg_buf *)substream->dma_private; -Then in :c:func:`snd_pcm_lib_malloc_pages()` call, the common SG-buffer +Then in the :c:func:`snd_pcm_lib_malloc_pages()` call, the common SG-buffer handler will allocate the non-contiguous kernel pages of the given size -and map them onto the virtually contiguous memory. The virtual pointer -is addressed in runtime->dma_area. The physical address +and map them as virtually contiguous memory. The virtual pointer +is addressed via runtime->dma_area. The physical address (``runtime->dma_addr``) is set to zero, because the buffer is physically non-contiguous. The physical address table is set up in ``sgbuf->table``. You can get the physical address at a certain offset @@ -3796,22 +3601,20 @@ Vmalloc'ed Buffers ------------------ It's possible to use a buffer allocated via :c:func:`vmalloc()`, for -example, for an intermediate buffer. In the recent version of kernel, -you can simply allocate it via standard -:c:func:`snd_pcm_lib_malloc_pages()` and co after setting up the -buffer preallocation with ``SNDRV_DMA_TYPE_VMALLOC`` type. - -:: +example, for an intermediate buffer. +You can simply allocate it via the standard +:c:func:`snd_pcm_lib_malloc_pages()` and co. after setting up the +buffer preallocation with ``SNDRV_DMA_TYPE_VMALLOC`` type:: snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_VMALLOC, NULL, 0, 0); -The NULL is passed to the device pointer argument, which indicates -that the default pages (GFP_KERNEL and GFP_HIGHMEM) will be +NULL is passed as the device pointer argument, which indicates +that default pages (GFP_KERNEL and GFP_HIGHMEM) will be allocated. -Also, note that zero is passed to both the size and the max size -arguments here. Since each vmalloc call should succeed at any time, +Also, note that zero is passed as both the size and the max size +argument here. Since each vmalloc call should succeed at any time, we don't need to pre-allocate the buffers like other continuous pages. @@ -3823,9 +3626,7 @@ useful for debugging. I recommend you set up proc files if you write a driver and want to get a running status or register dumps. The API is found in ``<sound/info.h>``. -To create a proc file, call :c:func:`snd_card_proc_new()`. - -:: +To create a proc file, call :c:func:`snd_card_proc_new()`:: struct snd_info_entry *entry; int err = snd_card_proc_new(card, "my-file", &entry); @@ -3841,28 +3642,22 @@ automatically in the card registration and release functions. When the creation is successful, the function stores a new instance in the pointer given in the third argument. It is initialized as a text proc file for read only. To use this proc file as a read-only text file -as it is, set the read callback with a private data via -:c:func:`snd_info_set_text_ops()`. - -:: +as-is, set the read callback with private data via +:c:func:`snd_info_set_text_ops()`:: snd_info_set_text_ops(entry, chip, my_proc_read); where the second argument (``chip``) is the private data to be used in -the callbacks. The third parameter specifies the read buffer size and +the callback. The third parameter specifies the read buffer size and the fourth (``my_proc_read``) is the callback function, which is -defined like - -:: +defined like:: static void my_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer); In the read callback, use :c:func:`snd_iprintf()` for output strings, which works just like normal :c:func:`printf()`. For -example, - -:: +example:: static void my_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) @@ -3873,28 +3668,22 @@ example, snd_iprintf(buffer, "Port = %ld\n", chip->port); } -The file permissions can be changed afterwards. As default, it's set as +The file permissions can be changed afterwards. By default, they are read only for all users. If you want to add write permission for the -user (root as default), do as follows: - -:: +user (root by default), do as follows:: entry->mode = S_IFREG | S_IRUGO | S_IWUSR; -and set the write buffer size and the callback - -:: +and set the write buffer size and the callback:: entry->c.text.write = my_proc_write; -For the write callback, you can use :c:func:`snd_info_get_line()` +In the write callback, you can use :c:func:`snd_info_get_line()` to get a text line, and :c:func:`snd_info_get_str()` to retrieve a string from the line. Some examples are found in ``core/oss/mixer_oss.c``, core/oss/and ``pcm_oss.c``. -For a raw-data proc-file, set the attributes as follows: - -:: +For a raw-data proc-file, set the attributes as follows:: static const struct snd_info_entry_ops my_file_io_ops = { .read = my_file_io_read, @@ -3906,14 +3695,13 @@ For a raw-data proc-file, set the attributes as follows: entry->size = 4096; entry->mode = S_IFREG | S_IRUGO; -For the raw data, ``size`` field must be set properly. This specifies +For raw data, ``size`` field must be set properly. This specifies the maximum size of the proc file access. The read/write callbacks of raw mode are more direct than the text mode. You need to use a low-level I/O functions such as -:c:func:`copy_from_user()` and :c:func:`copy_to_user()` to transfer the data. - -:: +:c:func:`copy_from_user()` and :c:func:`copy_to_user()` to transfer the +data:: static ssize_t my_file_io_read(struct snd_info_entry *entry, void *file_private_data, @@ -3938,12 +3726,11 @@ Power Management If the chip is supposed to work with suspend/resume functions, you need to add power-management code to the driver. The additional code for power-management should be ifdef-ed with ``CONFIG_PM``, or annotated -with __maybe_unused attribute; otherwise the compiler will complain -you. +with __maybe_unused attribute; otherwise the compiler will complain. If the driver *fully* supports suspend/resume that is, the device can be properly resumed to its state when suspend was called, you can set the -``SNDRV_PCM_INFO_RESUME`` flag in the pcm info field. Usually, this is +``SNDRV_PCM_INFO_RESUME`` flag in the PCM info field. Usually, this is possible when the registers of the chip can be safely saved and restored to RAM. If this is set, the trigger callback is called with ``SNDRV_PCM_TRIGGER_RESUME`` after the resume callback completes. @@ -3953,7 +3740,7 @@ is still possible, it's still worthy to implement suspend/resume callbacks. In such a case, applications would reset the status by calling :c:func:`snd_pcm_prepare()` and restart the stream appropriately. Hence, you can define suspend/resume callbacks below but -don't set ``SNDRV_PCM_INFO_RESUME`` info flag to the PCM. +don't set the ``SNDRV_PCM_INFO_RESUME`` info flag to the PCM. Note that the trigger with SUSPEND can always be called when :c:func:`snd_pcm_suspend_all()` is called, regardless of the @@ -3963,12 +3750,9 @@ behavior of :c:func:`snd_pcm_resume()`. (Thus, in theory, callback when no ``SNDRV_PCM_INFO_RESUME`` flag is set. But, it's better to keep it for compatibility reasons.) -In the earlier version of ALSA drivers, a common power-management layer -was provided, but it has been removed. The driver needs to define the +The driver needs to define the suspend/resume hooks according to the bus the device is connected to. In -the case of PCI drivers, the callbacks look like below: - -:: +the case of PCI drivers, the callbacks look like below:: static int __maybe_unused snd_my_suspend(struct device *dev) { @@ -3981,7 +3765,7 @@ the case of PCI drivers, the callbacks look like below: return 0; } -The scheme of the real suspend job is as follows. +The scheme of the real suspend job is as follows: 1. Retrieve the card and the chip data. @@ -3995,9 +3779,7 @@ The scheme of the real suspend job is as follows. 5. Stop the hardware if necessary. -A typical code would be like: - -:: +Typical code would look like:: static int __maybe_unused mychip_suspend(struct device *dev) { @@ -4016,7 +3798,7 @@ A typical code would be like: } -The scheme of the real resume job is as follows. +The scheme of the real resume job is as follows: 1. Retrieve the card and the chip data. @@ -4024,16 +3806,14 @@ The scheme of the real resume job is as follows. 3. Restore the saved registers if necessary. -4. Resume the mixer, e.g. calling :c:func:`snd_ac97_resume()`. +4. Resume the mixer, e.g. by calling :c:func:`snd_ac97_resume()`. 5. Restart the hardware (if any). 6. Call :c:func:`snd_power_change_state()` with ``SNDRV_CTL_POWER_D0`` to notify the processes. -A typical code would be like: - -:: +Typical code would look like:: static int __maybe_unused mychip_resume(struct pci_dev *pci) { @@ -4060,9 +3840,7 @@ been already suspended via its own PM ops calling OK, we have all callbacks now. Let's set them up. In the initialization of the card, make sure that you can get the chip data from the card instance, typically via ``private_data`` field, in case you created the -chip data individually. - -:: +chip data individually:: static int snd_mychip_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) @@ -4082,9 +3860,7 @@ chip data individually. } When you created the chip data with :c:func:`snd_card_new()`, it's -anyway accessible via ``private_data`` field. - -:: +anyway accessible via ``private_data`` field:: static int snd_mychip_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) @@ -4101,14 +3877,12 @@ anyway accessible via ``private_data`` field. .... } -If you need a space to save the registers, allocate the buffer for it +If you need space to save the registers, allocate the buffer for it here, too, since it would be fatal if you cannot allocate a memory in the suspend phase. The allocated buffer should be released in the corresponding destructor. -And next, set suspend/resume callbacks to the pci_driver. - -:: +And next, set suspend/resume callbacks to the pci_driver:: static SIMPLE_DEV_PM_OPS(snd_my_pm_ops, mychip_suspend, mychip_resume); @@ -4128,9 +3902,7 @@ have the ``index``, ``id`` and ``enable`` options. If the module supports multiple cards (usually up to 8 = ``SNDRV_CARDS`` cards), they should be arrays. The default initial values are defined -already as constants for easier programming: - -:: +already as constants for easier programming:: static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; @@ -4144,9 +3916,7 @@ The module parameters must be declared with the standard ``module_param()``, ``module_param_array()`` and :c:func:`MODULE_PARM_DESC()` macros. -The typical coding would be like below: - -:: +Typical code would look as below:: #define CARD_NAME "My Chip" @@ -4159,9 +3929,7 @@ The typical coding would be like below: Also, don't forget to define the module description and the license. Especially, the recent modprobe requires to define the -module license as GPL, etc., otherwise the system is shown as “tainted”. - -:: +module license as GPL, etc., otherwise the system is shown as “tainted”:: MODULE_DESCRIPTION("Sound driver for My Chip"); MODULE_LICENSE("GPL"); @@ -4224,32 +3992,36 @@ Driver with A Single Source File 1. Modify sound/pci/Makefile - Suppose you have a file xyz.c. Add the following two lines + Suppose you have a file xyz.c. Add the following two lines:: -:: - - snd-xyz-objs := xyz.o - obj-$(CONFIG_SND_XYZ) += snd-xyz.o + snd-xyz-objs := xyz.o + obj-$(CONFIG_SND_XYZ) += snd-xyz.o 2. Create the Kconfig entry - Add the new entry of Kconfig for your xyz driver. config SND_XYZ - tristate "Foobar XYZ" depends on SND select SND_PCM help Say Y here - to include support for Foobar XYZ soundcard. To compile this driver - as a module, choose M here: the module will be called snd-xyz. the - line, select SND_PCM, specifies that the driver xyz supports PCM. In - addition to SND_PCM, the following components are supported for - select command: SND_RAWMIDI, SND_TIMER, SND_HWDEP, - SND_MPU401_UART, SND_OPL3_LIB, SND_OPL4_LIB, SND_VX_LIB, - SND_AC97_CODEC. Add the select command for each supported - component. + Add the new entry of Kconfig for your xyz driver:: + + config SND_XYZ + tristate "Foobar XYZ" + depends on SND + select SND_PCM + help + Say Y here to include support for Foobar XYZ soundcard. + To compile this driver as a module, choose M here: + the module will be called snd-xyz. + +The line ``select SND_PCM`` specifies that the driver xyz supports PCM. +In addition to SND_PCM, the following components are supported for +select command: SND_RAWMIDI, SND_TIMER, SND_HWDEP, SND_MPU401_UART, +SND_OPL3_LIB, SND_OPL4_LIB, SND_VX_LIB, SND_AC97_CODEC. +Add the select command for each supported component. - Note that some selections imply the lowlevel selections. For example, - PCM includes TIMER, MPU401_UART includes RAWMIDI, AC97_CODEC - includes PCM, and OPL3_LIB includes HWDEP. You don't need to give - the lowlevel selections again. +Note that some selections imply the lowlevel selections. For example, +PCM includes TIMER, MPU401_UART includes RAWMIDI, AC97_CODEC +includes PCM, and OPL3_LIB includes HWDEP. You don't need to give +the lowlevel selections again. - For the details of Kconfig script, refer to the kbuild documentation. +For the details of Kconfig script, refer to the kbuild documentation. Drivers with Several Source Files --------------------------------- @@ -4258,16 +4030,12 @@ Suppose that the driver snd-xyz have several source files. They are located in the new subdirectory, sound/pci/xyz. 1. Add a new directory (``sound/pci/xyz``) in ``sound/pci/Makefile`` - as below + as below:: -:: - - obj-$(CONFIG_SND) += sound/pci/xyz/ + obj-$(CONFIG_SND) += sound/pci/xyz/ -2. Under the directory ``sound/pci/xyz``, create a Makefile - -:: +2. Under the directory ``sound/pci/xyz``, create a Makefile:: snd-xyz-objs := xyz.o abc.o def.o obj-$(CONFIG_SND_XYZ) += snd-xyz.o diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h index 49200ec26dc4..c495c6d5fbe0 100644 --- a/include/sound/ac97_codec.h +++ b/include/sound/ac97_codec.h @@ -335,6 +335,9 @@ static inline int snd_ac97_update_power(struct snd_ac97 *ac97, int reg, #ifdef CONFIG_PM void snd_ac97_suspend(struct snd_ac97 *ac97); void snd_ac97_resume(struct snd_ac97 *ac97); +#else +static inline void snd_ac97_suspend(struct snd_ac97 *ac97) {} +static inline void snd_ac97_resume(struct snd_ac97 *ac97) {} #endif int snd_ac97_reset(struct snd_ac97 *ac97, bool try_warm, unsigned int id, unsigned int id_mask); diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index 39787fecc8d9..8fe80dcee71b 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -25,13 +25,9 @@ /* ------------------- DEFINES -------------------- */ #define EMUPAGESIZE 4096 -#define MAXREQVOICES 8 #define MAXPAGES0 4096 /* 32 bit mode */ #define MAXPAGES1 8192 /* 31 bit mode */ -#define RESERVED 0 -#define NUM_MIDI 16 #define NUM_G 64 /* use all channels */ -#define NUM_FXSENDS 4 #define NUM_EFX_PLAYBACK 16 /* FIXME? - according to the OSS driver the EMU10K1 needs a 29 bit DMA mask */ @@ -39,7 +35,6 @@ #define AUDIGY_DMA_MASK 0xffffffffUL /* 32bit mode */ #define TMEMSIZE 256*1024 -#define TMEMSIZEREG 4 #define IP_TO_CP(ip) ((ip == 0) ? 0 : (((0x00001000uL | (ip & 0x00000FFFL)) << (((ip >> 12) & 0x000FL) + 4)) & 0xFFFF0000uL)) @@ -66,8 +61,8 @@ /* the relevant bits and zero to the other bits */ #define IPR_P16V 0x80000000 /* Bit set when the CA0151 P16V chip wishes to interrupt */ -#define IPR_GPIOMSG 0x20000000 /* GPIO message interrupt (RE'd, still not sure - which INTE bits enable it) */ +#define IPR_WATERMARK_REACHED 0x40000000 +#define IPR_A_GPIO 0x20000000 /* GPIO input pin change */ /* The next two interrupts are for the midi port on the Audigy Drive (A_MPU1) */ #define IPR_A_MIDITRANSBUFEMPTY2 0x10000000 /* MIDI UART transmit buffer empty */ @@ -97,9 +92,9 @@ #define IPR_CHANNELLOOP 0x00000040 /* Channel (half) loop interrupt(s) pending */ #define IPR_CHANNELNUMBERMASK 0x0000003f /* When IPR_CHANNELLOOP is set, indicates the */ /* highest set channel in CLIPL, CLIPH, HLIPL, */ - /* or HLIPH. When IP is written with CL set, */ + /* or HLIPH. When IPR is written with CL set, */ /* the bit in H/CLIPL or H/CLIPH corresponding */ - /* to the CIN value written will be cleared. */ + /* to the CN value written will be cleared. */ #define INTE 0x0c /* Interrupt enable register */ #define INTE_VIRTUALSB_MASK 0xc0000000 /* Virtual Soundblaster I/O port capture */ @@ -127,10 +122,14 @@ /* behavior and possibly random segfaults and */ /* lockups if enabled. */ +#define INTE_A_GPIOENABLE 0x00040000 /* Enable GPIO input change interrupts */ + /* The next two interrupts are for the midi port on the Audigy Drive (A_MPU1) */ #define INTE_A_MIDITXENABLE2 0x00020000 /* Enable MIDI transmit-buffer-empty interrupts */ #define INTE_A_MIDIRXENABLE2 0x00010000 /* Enable MIDI receive-buffer-empty interrupts */ +#define INTE_A_SPDIF_BUFFULL_ENABLE 0x00008000 +#define INTE_A_SPDIF_HALFBUFFULL_ENABLE 0x00004000 #define INTE_SAMPLERATETRACKER 0x00002000 /* Enable sample rate tracker interrupts */ /* NOTE: This bit must always be enabled */ @@ -151,9 +150,10 @@ #define WC 0x10 /* Wall Clock register */ #define WC_SAMPLECOUNTER_MASK 0x03FFFFC0 /* Sample periods elapsed since reset */ #define WC_SAMPLECOUNTER 0x14060010 -#define WC_CURRENTCHANNEL 0x0000003F /* Channel [0..63] currently being serviced */ +#define WC_CURRENTCHANNEL_MASK 0x0000003F /* Channel [0..63] currently being serviced */ /* NOTE: Each channel takes 1/64th of a sample */ /* period to be serviced. */ +#define WC_CURRENTCHANNEL 0x06000010 #define HCFG 0x14 /* Hardware config register */ /* NOTE: There is no reason to use the legacy */ @@ -180,6 +180,7 @@ #define HCFG_CODECFORMAT_MASK 0x00030000 /* CODEC format */ /* Specific to Alice2, CA0102 */ + #define HCFG_CODECFORMAT_AC97_1 0x00000000 /* AC97 CODEC format -- Ver 1.03 */ #define HCFG_CODECFORMAT_AC97_2 0x00010000 /* AC97 CODEC format -- Ver 2.1 */ #define HCFG_AUTOMUTE_ASYNC 0x00008000 /* When set, the async sample rate convertors */ @@ -200,9 +201,8 @@ /* I2S format input */ /* Rest of HCFG 0x0000000f same as below. LOCKSOUNDCACHE etc. */ - - /* Older chips */ + #define HCFG_CODECFORMAT_AC97 0x00000000 /* AC97 CODEC format -- Primary Output */ #define HCFG_CODECFORMAT_I2S 0x00010000 /* I2S CODEC format -- Secondary (Rear) Output */ #define HCFG_GPINPUT0 0x00004000 /* External pin112 */ @@ -238,7 +238,7 @@ /* Should be set to 1 when the EMU10K1 is */ /* completely initialized. */ -//For Audigy, MPU port move to 0x70-0x74 ptr register +// On Audigy, the MPU port moved to the 0x70-0x74 ptr registers #define MUDATA 0x18 /* MPU401 data register (8 bits) */ @@ -251,11 +251,17 @@ #define MUSTAT_IRDYN 0x80 /* 0 = MIDI data or command ACK */ #define MUSTAT_ORDYN 0x40 /* 0 = MUDATA can accept a command or data */ -#define A_IOCFG 0x18 /* GPIO on Audigy card (16bits) */ -#define A_GPINPUT_MASK 0xff00 +#define A_GPIO 0x18 /* GPIO on Audigy card (16bits) */ +#define A_GPINPUT_MASK 0xff00 /* Alice/2 has 8 input pins */ +#define A3_GPINPUT_MASK 0x3f00 /* ... while Tina/2 has only 6 */ #define A_GPOUTPUT_MASK 0x00ff +// The GPIO port is used for I/O config on Sound Blasters; +// card-specific info can be found in the emu_chip_details table. +// On E-MU cards the port is used as the interface to the FPGA. + // Audigy output/GPIO stuff taken from the kX drivers +#define A_IOCFG A_GPIO #define A_IOCFG_GPOUT0 0x0044 /* analog/digital */ #define A_IOCFG_DISABLE_ANALOG 0x0040 /* = 'enable' for Audigy2 (chiprev=4) */ #define A_IOCFG_ENABLE_DIGITAL 0x0004 @@ -271,19 +277,12 @@ #define A_IOCFG_REAR_JACK 0x8000 #define A_IOCFG_PHONES_JACK 0x0100 /* LiveDrive */ -/* outputs: - * for audigy2 platinum: 0xa00 - * for a2 platinum ex: 0x1c00 - * for a1 platinum: 0x0 - */ - #define TIMER 0x1a /* Timer terminal count register */ /* NOTE: After the rate is changed, a maximum */ /* of 1024 sample periods should be allowed */ /* before the new rate is guaranteed accurate. */ -#define TIMER_RATE_MASK 0x000003ff /* Timer interrupt rate in sample periods */ +#define TIMER_RATE_MASK 0x03ff /* Timer interrupt rate in sample periods */ /* 0 == 1024 periods, [1..4] are not useful */ -#define TIMER_RATE 0x0a00001a #define AC97DATA 0x1c /* AC97 register set data register (16 bit) */ @@ -317,7 +316,7 @@ /* 0x00000000 2-channel output. */ /* 0x00000200 8-channel output. */ /* 0x00000004 pauses stream/irq fail. */ - /* Rest of bits no nothing to sound output */ + /* Rest of bits do nothing to sound output */ /* bit 0: Enable P16V audio. * bit 1: Lock P16V record memory cache. * bit 2: Lock P16V playback memory cache. @@ -331,6 +330,7 @@ */ #define IPR3 0x38 /* Cdif interrupt pending register */ #define INTE3 0x3c /* Cdif interrupt enable register. */ + /************************************************************************************************/ /* PCI function 1 registers, address = <val> + PCIBASE1 */ /************************************************************************************************/ @@ -349,11 +349,38 @@ #define JOYSTICK_BUTTONS 0x0f /* Joystick button data */ #define JOYSTICK_COMPARATOR 0xf0 /* Joystick comparator data */ - /********************************************************************************************************/ /* Emu10k1 pointer-offset register set, accessed through the PTR and DATA registers */ /********************************************************************************************************/ +// No official documentation was released for EMU10K1, but some info +// about playback can be extrapolated from the EMU8K documents: +// "AWE32/EMU8000 Programmer’s Guide" (emu8kpgm.pdf) - registers +// "AWE32 Developer's Information Pack" (adip301.pdf) - high-level view + +// The short version: +// - The engine has 64 playback channels, also called voices. The channels +// operate independently, except when paired for stereo (see below). +// - PCM samples are fetched into the cache; see description of CD0 below. +// - Samples are consumed at the rate CPF_CURRENTPITCH. +// - 8-bit samples are transformed upon use: cooked = (raw ^ 0x80) << 8 +// - 8 samples are read at CCR_READADDRESS:CPF_FRACADDRESS and interpolated +// according to CCCA_INTERPROM_*. With CCCA_INTERPROM_0 selected and a zero +// CPF_FRACADDRESS, this results in CCR_READADDRESS[3] being used verbatim. +// - The value is multiplied by CVCF_CURRENTVOL. +// - The value goes through a filter with cutoff CVCF_CURRENTFILTER; +// delay stages Z1 and Z2. +// - The value is added by so-called `sends` to 4 (EMU10K1) / 8 (EMU10K2) +// of the 16 (EMU10K1) / 64 (EMU10K2) FX bus accumulators via FXRT*, +// multiplied by a per-send amount (*_FXSENDAMOUNT_*). +// The scaling of the send amounts is exponential-ish. +// - The DSP has a go at FXBUS* and outputs the values to EXTOUT* or EMU32OUT*. +// - The pitch, volume, and filter cutoff can be modulated by two envelope +// engines and two low frequency oscillators. +// - To avoid abrupt changes to the parameters (which may cause audible +// distortion), the modulation engine sets the target registers, towards +// which the current registers "swerve" gradually. + #define CPF 0x00 /* Current pitch and fraction register */ #define CPF_CURRENTPITCH_MASK 0xffff0000 /* Current pitch (linear, 0x4000 == unity pitch shift) */ #define CPF_CURRENTPITCH 0x10100000 @@ -393,7 +420,7 @@ #define PSST_LOOPSTARTADDR_MASK 0x00ffffff /* Loop start address of the specified channel */ #define PSST_LOOPSTARTADDR 0x18000006 -#define DSL 0x07 /* Send D amount and loop start address register */ +#define DSL 0x07 /* Send D amount and loop end address register */ #define DSL_FXSENDAMOUNT_D_MASK 0xff000000 /* Linear level of channel output sent to FX send bus D */ #define DSL_FXSENDAMOUNT_D 0x08180007 @@ -402,8 +429,9 @@ #define DSL_LOOPENDADDR 0x18000007 #define CCCA 0x08 /* Filter Q, interp. ROM, byte size, cur. addr register */ -#define CCCA_RESONANCE 0xf0000000 /* Lowpass filter resonance (Q) height */ -#define CCCA_INTERPROMMASK 0x0e000000 /* Selects passband of interpolation ROM */ +#define CCCA_RESONANCE_MASK 0xf0000000 /* Lowpass filter resonance (Q) height */ +#define CCCA_RESONANCE 0x041c0008 +#define CCCA_INTERPROM_MASK 0x0e000000 /* Selects passband of interpolation ROM */ /* 1 == full band, 7 == lowpass */ /* ROM 0 is used when pitch shifting downward or less */ /* then 3 semitones upward. Increasingly higher ROM */ @@ -418,25 +446,28 @@ #define CCCA_INTERPROM_6 0x0c000000 /* Select interpolation ROM 6 */ #define CCCA_INTERPROM_7 0x0e000000 /* Select interpolation ROM 7 */ #define CCCA_8BITSELECT 0x01000000 /* 1 = Sound memory for this channel uses 8-bit samples */ + /* 8-bit samples are unsigned, 16-bit ones signed */ #define CCCA_CURRADDR_MASK 0x00ffffff /* Current address of the selected channel */ #define CCCA_CURRADDR 0x18000008 #define CCR 0x09 /* Cache control register */ #define CCR_CACHEINVALIDSIZE 0x07190009 -#define CCR_CACHEINVALIDSIZE_MASK 0xfe000000 /* Number of invalid samples cache for this channel */ +#define CCR_CACHEINVALIDSIZE_MASK 0xfe000000 /* Number of invalid samples before the read address */ #define CCR_CACHELOOPFLAG 0x01000000 /* 1 = Cache has a loop service pending */ #define CCR_INTERLEAVEDSAMPLES 0x00800000 /* 1 = A cache service will fetch interleaved samples */ + /* Auto-set from CPF_STEREO_MASK */ #define CCR_WORDSIZEDSAMPLES 0x00400000 /* 1 = A cache service will fetch word sized samples */ + /* Auto-set from CCCA_8BITSELECT */ #define CCR_READADDRESS 0x06100009 -#define CCR_READADDRESS_MASK 0x003f0000 /* Location of cache just beyond current cache service */ +#define CCR_READADDRESS_MASK 0x003f0000 /* Next cached sample to play */ #define CCR_LOOPINVALSIZE 0x0000fe00 /* Number of invalid samples in cache prior to loop */ /* NOTE: This is valid only if CACHELOOPFLAG is set */ #define CCR_LOOPFLAG 0x00000100 /* Set for a single sample period when a loop occurs */ -#define CCR_CACHELOOPADDRHI 0x000000ff /* DSL_LOOPSTARTADDR's hi byte if CACHELOOPFLAG is set */ +#define CCR_CACHELOOPADDRHI 0x000000ff /* CLP_LOOPSTARTADDR's hi byte if CACHELOOPFLAG is set */ #define CLP 0x0a /* Cache loop register (valid if CCR_CACHELOOPFLAG = 1) */ /* NOTE: This register is normally not used */ -#define CLP_CACHELOOPADDR 0x0000ffff /* Cache loop address (DSL_LOOPSTARTADDR [0..15]) */ +#define CLP_CACHELOOPADDR 0x0000ffff /* Cache loop address low word */ #define FXRT 0x0b /* Effects send routing register */ /* NOTE: It is illegal to assign the same routing to */ @@ -446,9 +477,7 @@ #define FXRT_CHANNELC 0x0f000000 /* Effects send bus number for channel's effects send C */ #define FXRT_CHANNELD 0xf0000000 /* Effects send bus number for channel's effects send D */ -#define A_HR 0x0b /* High Resolution. 24bit playback from host to DSP. */ #define MAPA 0x0c /* Cache map A */ - #define MAPB 0x0d /* Cache map B */ #define MAP_PTE_MASK0 0xfffff000 /* The 20 MSBs of the PTE indexed by the PTI */ @@ -457,22 +486,22 @@ #define MAP_PTE_MASK1 0xffffe000 /* The 19 MSBs of the PTE indexed by the PTI */ #define MAP_PTI_MASK1 0x00001fff /* The 13 bit index to one of the 8192 PTE dwords */ -/* 0x0e, 0x0f: Not used */ +/* 0x0e, 0x0f: Internal state, at least on Audigy */ #define ENVVOL 0x10 /* Volume envelope register */ #define ENVVOL_MASK 0x0000ffff /* Current value of volume envelope state variable */ /* 0x8000-n == 666*n usec delay */ #define ATKHLDV 0x11 /* Volume envelope hold and attack register */ -#define ATKHLDV_PHASE0 0x00008000 /* 0 = Begin attack phase */ +#define ATKHLDV_PHASE0_MASK 0x00008000 /* 0 = Begin attack phase */ #define ATKHLDV_HOLDTIME_MASK 0x00007f00 /* Envelope hold time (127-n == n*88.2msec) */ #define ATKHLDV_ATTACKTIME_MASK 0x0000007f /* Envelope attack time, log encoded */ /* 0 = infinite, 1 = 10.9msec, ... 0x7f = 5.5msec */ #define DCYSUSV 0x12 /* Volume envelope sustain and decay register */ -#define DCYSUSV_PHASE1_MASK 0x00008000 /* 0 = Begin attack phase, 1 = begin release phase */ +#define DCYSUSV_PHASE1_MASK 0x00008000 /* 0 = Begin decay phase, 1 = begin release phase */ #define DCYSUSV_SUSTAINLEVEL_MASK 0x00007f00 /* 127 = full, 0 = off, 0.75dB increments */ -#define DCYSUSV_CHANNELENABLE_MASK 0x00000080 /* 1 = Inhibit envelope engine from writing values in */ +#define DCYSUSV_CHANNELENABLE_MASK 0x00000080 /* 0 = Inhibit envelope engine from writing values in */ /* this channel and from writing to pitch, filter and */ /* volume targets. */ #define DCYSUSV_DECAYTIME_MASK 0x0000007f /* Volume envelope decay time, log encoded */ @@ -487,13 +516,13 @@ /* 0x8000-n == 666*n usec delay */ #define ATKHLDM 0x15 /* Modulation envelope hold and attack register */ -#define ATKHLDM_PHASE0 0x00008000 /* 0 = Begin attack phase */ +#define ATKHLDM_PHASE0_MASK 0x00008000 /* 0 = Begin attack phase */ #define ATKHLDM_HOLDTIME 0x00007f00 /* Envelope hold time (127-n == n*42msec) */ #define ATKHLDM_ATTACKTIME 0x0000007f /* Envelope attack time, log encoded */ /* 0 = infinite, 1 = 11msec, ... 0x7f = 5.5msec */ #define DCYSUSM 0x16 /* Modulation envelope decay and sustain register */ -#define DCYSUSM_PHASE1_MASK 0x00008000 /* 0 = Begin attack phase, 1 = begin release phase */ +#define DCYSUSM_PHASE1_MASK 0x00008000 /* 0 = Begin decay phase, 1 = begin release phase */ #define DCYSUSM_SUSTAINLEVEL_MASK 0x00007f00 /* 127 = full, 0 = off, 0.75dB increments */ #define DCYSUSM_DECAYTIME_MASK 0x0000007f /* Envelope decay time, log encoded */ /* 0 = 43.7msec, 1 = 21.8msec, 0x7f = 22msec */ @@ -515,7 +544,6 @@ #define IFATN_ATTENUATION_MASK 0x000000ff /* Initial attenuation in 0.375dB steps */ #define IFATN_ATTENUATION 0x08000019 - #define PEFE 0x1a /* Pitch envelope and filter envelope amount register */ #define PEFE_PITCHAMOUNT_MASK 0x0000ff00 /* Pitch envlope amount */ /* Signed 2's complement, +/- one octave peak extremes */ @@ -523,19 +551,19 @@ #define PEFE_FILTERAMOUNT_MASK 0x000000ff /* Filter envlope amount */ /* Signed 2's complement, +/- six octaves peak extremes */ #define PEFE_FILTERAMOUNT 0x0800001a + #define FMMOD 0x1b /* Vibrato/filter modulation from LFO register */ #define FMMOD_MODVIBRATO 0x0000ff00 /* Vibrato LFO modulation depth */ /* Signed 2's complement, +/- one octave extremes */ #define FMMOD_MOFILTER 0x000000ff /* Filter LFO modulation depth */ /* Signed 2's complement, +/- three octave extremes */ - #define TREMFRQ 0x1c /* Tremolo amount and modulation LFO frequency register */ #define TREMFRQ_DEPTH 0x0000ff00 /* Tremolo depth */ /* Signed 2's complement, with +/- 12dB extremes */ - #define TREMFRQ_FREQUENCY 0x000000ff /* Tremolo LFO frequency */ /* ??Hz steps, maximum of ?? Hz. */ + #define FM2FRQ2 0x1d /* Vibrato amount and vibrato LFO frequency register */ #define FM2FRQ2_DEPTH 0x0000ff00 /* Vibrato LFO vibrato depth */ /* Signed 2's complement, +/- one octave extremes */ @@ -603,20 +631,6 @@ /* is 16bit, 48KHz only. All 32 channels can be enabled */ /* simultaneously. */ -#define FXWC_DEFAULTROUTE_C (1<<0) /* left emu out? */ -#define FXWC_DEFAULTROUTE_B (1<<1) /* right emu out? */ -#define FXWC_DEFAULTROUTE_A (1<<12) -#define FXWC_DEFAULTROUTE_D (1<<13) -#define FXWC_ADCLEFT (1<<18) -#define FXWC_CDROMSPDIFLEFT (1<<18) -#define FXWC_ADCRIGHT (1<<19) -#define FXWC_CDROMSPDIFRIGHT (1<<19) -#define FXWC_MIC (1<<20) -#define FXWC_ZOOMLEFT (1<<20) -#define FXWC_ZOOMRIGHT (1<<21) -#define FXWC_SPDIFLEFT (1<<22) /* 0x00400000 */ -#define FXWC_SPDIFRIGHT (1<<23) /* 0x00800000 */ - #define A_TBLSZ 0x43 /* Effects Tank Internal Table Size. Only low byte or register used */ #define TCBS 0x44 /* Tank cache buffer size register */ @@ -639,7 +653,7 @@ #define FXBA 0x47 /* FX Buffer Address */ #define FXBA_MASK 0xfffff000 /* 20 bit base address */ -#define A_HWM 0x48 /* High PCI Water Mark - word access, defaults to 3f */ +#define A_HWM 0x48 /* High PCI Water Mark - word access, defaults to 3f */ #define MICBS 0x49 /* Microphone buffer size register */ @@ -647,9 +661,7 @@ #define FXBS 0x4b /* FX buffer size register */ -/* register: 0x4c..4f: ffff-ffff current amounts, per-channel */ - -/* The following mask values define the size of the ADC, MIX and FX buffers in bytes */ +/* The following mask values define the size of the ADC, MIC and FX buffers in bytes */ #define ADCBS_BUFSIZE_NONE 0x00000000 #define ADCBS_BUFSIZE_384 0x00000001 #define ADCBS_BUFSIZE_448 0x00000002 @@ -683,38 +695,27 @@ #define ADCBS_BUFSIZE_57344 0x0000001e #define ADCBS_BUFSIZE_65536 0x0000001f -/* Current Send B, A Amounts */ -#define A_CSBA 0x4c - -/* Current Send D, C Amounts */ -#define A_CSDC 0x4d - -/* Current Send F, E Amounts */ -#define A_CSFE 0x4e - -/* Current Send H, G Amounts */ -#define A_CSHG 0x4f +#define A_CSBA 0x4c /* FX send B & A current amounts */ +#define A_CSDC 0x4d /* FX send D & C current amounts */ +#define A_CSFE 0x4e /* FX send F & E current amounts */ +#define A_CSHG 0x4f /* FX send H & G current amounts */ +// NOTE: 0x50,51,52: 64-bit (split over voices 0 & 1) +#define CDCS 0x50 /* CD-ROM digital channel status register */ -#define CDCS 0x50 /* CD-ROM digital channel status register */ +#define GPSCS 0x51 /* General Purpose SPDIF channel status register */ -#define GPSCS 0x51 /* General Purpose SPDIF channel status register*/ +// Corresponding EMU10K1_DBG_* constants are in the public header +#define DBG 0x52 -#define DBG 0x52 /* DO NOT PROGRAM THIS REGISTER!!! MAY DESTROY CHIP */ +#define A_SPSC 0x52 /* S/PDIF Input C Channel Status */ -/* S/PDIF Input C Channel Status */ -#define A_SPSC 0x52 +#define REG53 0x53 /* DO NOT PROGRAM THIS REGISTER!!! MAY DESTROY CHIP */ -#define REG53 0x53 /* DO NOT PROGRAM THIS REGISTER!!! MAY DESTROY CHIP */ +// Corresponding A_DBG_* constants are in the public header +#define A_DBG 0x53 -#define A_DBG 0x53 -#define A_DBG_SINGLE_STEP 0x00020000 /* Set to zero to start dsp */ -#define A_DBG_ZC 0x40000000 /* zero tram counter */ -#define A_DBG_STEP_ADDR 0x000003ff -#define A_DBG_SATURATION_OCCURED 0x20000000 -#define A_DBG_SATURATION_ADDR 0x0ffc0000 - -// NOTE: 0x54,55,56: 64-bit +// NOTE: 0x54,55,56: 64-bit (split over voices 0 & 1) #define SPCS0 0x54 /* SPDIF output Channel Status 0 register */ #define SPCS1 0x55 /* SPDIF output Channel Status 1 register */ @@ -747,17 +748,14 @@ /* 0x57: Not used */ -/* The 32-bit CLIx and SOLx registers all have one bit per channel control/status */ +/* The 32-bit CLIx and SOLEx registers all have one bit per channel control/status */ #define CLIEL 0x58 /* Channel loop interrupt enable low register */ - #define CLIEH 0x59 /* Channel loop interrupt enable high register */ #define CLIPL 0x5a /* Channel loop interrupt pending low register */ - #define CLIPH 0x5b /* Channel loop interrupt pending high register */ #define SOLEL 0x5c /* Stop on loop enable low register */ - #define SOLEH 0x5d /* Stop on loop enable high register */ #define SPBYPASS 0x5e /* SPDIF BYPASS mode register */ @@ -767,13 +765,12 @@ #define SPBYPASS_FORMAT 0x00000f00 /* If 1, SPDIF XX uses 24 bit, if 0 - 20 bit */ #define AC97SLOT 0x5f /* additional AC97 slots enable bits */ -#define AC97SLOT_REAR_RIGHT 0x01 /* Rear left */ -#define AC97SLOT_REAR_LEFT 0x02 /* Rear right */ -#define AC97SLOT_CNTR 0x10 /* Center enable */ -#define AC97SLOT_LFE 0x20 /* LFE enable */ +#define AC97SLOT_REAR_RIGHT 0x01 /* Rear left */ +#define AC97SLOT_REAR_LEFT 0x02 /* Rear right */ +#define AC97SLOT_CNTR 0x10 /* Center enable */ +#define AC97SLOT_LFE 0x20 /* LFE enable */ -/* PCB Revision */ -#define A_PCB 0x5f +#define A_PCB 0x5f /* PCB Revision */ // NOTE: 0x60,61,62: 64-bit #define CDSRCS 0x60 /* CD-ROM Sample Rate Converter status register */ @@ -813,27 +810,21 @@ #define FXIDX_MASK 0x0000ffff /* 16-bit value */ #define FXIDX_IDX 0x10000065 -/* The 32-bit HLIx and HLIPx registers all have one bit per channel control/status */ +/* The 32-bit HLIEx and HLIPx registers all have one bit per channel control/status */ #define HLIEL 0x66 /* Channel half loop interrupt enable low register */ - #define HLIEH 0x67 /* Channel half loop interrupt enable high register */ #define HLIPL 0x68 /* Channel half loop interrupt pending low register */ - #define HLIPH 0x69 /* Channel half loop interrupt pending high register */ -/* S/PDIF Host Record Index (bypasses SRC) */ -#define A_SPRI 0x6a -/* S/PDIF Host Record Address */ -#define A_SPRA 0x6b -/* S/PDIF Host Record Control */ -#define A_SPRC 0x6c -/* Delayed Interrupt Counter & Enable */ -#define A_DICE 0x6d -/* Tank Table Base */ -#define A_TTB 0x6e -/* Tank Delay Offset */ -#define A_TDOF 0x6f +#define A_SPRI 0x6a /* S/PDIF Host Record Index (bypasses SRC) */ +#define A_SPRA 0x6b /* S/PDIF Host Record Address */ +#define A_SPRC 0x6c /* S/PDIF Host Record Control */ + +#define A_DICE 0x6d /* Delayed Interrupt Counter & Enable */ + +#define A_TTB 0x6e /* Tank Table Base */ +#define A_TDOF 0x6f /* Tank Delay Offset */ /* This is the MPU port on the card (via the game port) */ #define A_MUDATA1 0x70 @@ -846,48 +837,53 @@ #define A_MUSTAT2 A_MUCMD2 /* The next two are the Audigy equivalent of FXWC */ -/* the Audigy can record any output (16bit, 48kHz, up to 64 channel simultaneously) */ +/* the Audigy can record any output (16bit, 48kHz, up to 64 channels simultaneously) */ /* Each bit selects a channel for recording */ #define A_FXWC1 0x74 /* Selects 0x7f-0x60 for FX recording */ #define A_FXWC2 0x75 /* Selects 0x9f-0x80 for FX recording */ -/* Extended Hardware Control */ -#define A_SPDIF_SAMPLERATE 0x76 /* Set the sample rate of SPDIF output */ -#define A_SAMPLE_RATE 0x76 /* Various sample rate settings. */ -#define A_SAMPLE_RATE_NOT_USED 0x0ffc111e /* Bits that are not used and cannot be set. */ -#define A_SAMPLE_RATE_UNKNOWN 0xf0030001 /* Bits that can be set, but have unknown use. */ +#define A_EHC 0x76 /* Extended Hardware Control */ + +#define A_SPDIF_SAMPLERATE A_EHC /* Set the sample rate of SPDIF output */ #define A_SPDIF_RATE_MASK 0x000000e0 /* Any other values for rates, just use 48000 */ -#define A_SPDIF_48000 0x00000000 +#define A_SPDIF_48000 0x00000000 /* kX calls this BYPASS */ #define A_SPDIF_192000 0x00000020 #define A_SPDIF_96000 0x00000040 #define A_SPDIF_44100 0x00000080 +#define A_SPDIF_MUTED 0x000000c0 #define A_I2S_CAPTURE_RATE_MASK 0x00000e00 /* This sets the capture PCM rate, but it is */ -#define A_I2S_CAPTURE_48000 0x00000000 /* unclear if this sets the ADC rate as well. */ -#define A_I2S_CAPTURE_192000 0x00000200 -#define A_I2S_CAPTURE_96000 0x00000400 -#define A_I2S_CAPTURE_44100 0x00000800 - -#define A_PCM_RATE_MASK 0x0000e000 /* This sets the playback PCM rate on the P16V */ -#define A_PCM_48000 0x00000000 -#define A_PCM_192000 0x00002000 -#define A_PCM_96000 0x00004000 -#define A_PCM_44100 0x00008000 - -/* I2S0 Sample Rate Tracker Status */ -#define A_SRT3 0x77 - -/* I2S1 Sample Rate Tracker Status */ -#define A_SRT4 0x78 - -/* I2S2 Sample Rate Tracker Status */ -#define A_SRT5 0x79 +#define A_I2S_CAPTURE_RATE 0x03090076 /* unclear if this sets the ADC rate as well. */ +#define A_I2S_CAPTURE_48000 0x0 +#define A_I2S_CAPTURE_192000 0x1 +#define A_I2S_CAPTURE_96000 0x2 +#define A_I2S_CAPTURE_44100 0x4 + +#define A_EHC_SRC48_MASK 0x0000e000 /* This sets the playback PCM rate on the P16V */ +#define A_EHC_SRC48_BYPASS 0x00000000 +#define A_EHC_SRC48_192 0x00002000 +#define A_EHC_SRC48_96 0x00004000 +#define A_EHC_SRC48_44 0x00008000 +#define A_EHC_SRC48_MUTED 0x0000c000 + +#define A_EHC_P17V_TVM 0x00000001 /* Tank virtual memory mode */ +#define A_EHC_P17V_SEL0_MASK 0x00030000 /* Aka A_EHC_P16V_PB_RATE; 00: 48, 01: 44.1, 10: 96, 11: 192 */ +#define A_EHC_P17V_SEL1_MASK 0x000c0000 +#define A_EHC_P17V_SEL2_MASK 0x00300000 +#define A_EHC_P17V_SEL3_MASK 0x00c00000 + +#define A_EHC_ASYNC_BYPASS 0x80000000 + +#define A_SRT3 0x77 /* I2S0 Sample Rate Tracker Status */ +#define A_SRT4 0x78 /* I2S1 Sample Rate Tracker Status */ +#define A_SRT5 0x79 /* I2S2 Sample Rate Tracker Status */ /* - default to 0x01080000 on my audigy 2 ZS --rlrevell */ -/* Tank Table DMA Address */ -#define A_TTDA 0x7a -/* Tank Table DMA Data */ -#define A_TTDD 0x7b +#define A_SRT_ESTSAMPLERATE 0x001fffff +#define A_SRT_RATELOCKED 0x01000000 + +#define A_TTDA 0x7a /* Tank Table DMA Address */ +#define A_TTDD 0x7b /* Tank Table DMA Data */ #define A_FXRT2 0x7c #define A_FXRT_CHANNELE 0x0000003f /* Effects send bus number for channel's effects send E */ @@ -900,6 +896,7 @@ #define A_FXSENDAMOUNT_F_MASK 0x00FF0000 #define A_FXSENDAMOUNT_G_MASK 0x0000FF00 #define A_FXSENDAMOUNT_H_MASK 0x000000FF + /* 0x7c, 0x7e "high bit is used for filtering" */ /* The send amounts for this one are the same as used with the emu10k1 */ @@ -910,55 +907,56 @@ #define A_FXRT_CHANNELD 0x3f000000 /* 0x7f: Not used */ -/* Each FX general purpose register is 32 bits in length, all bits are used */ -#define FXGPREGBASE 0x100 /* FX general purpose registers base */ -#define A_FXGPREGBASE 0x400 /* Audigy GPRs, 0x400 to 0x5ff */ - -#define A_TANKMEMCTLREGBASE 0x100 /* Tank memory control registers base - only for Audigy */ -#define A_TANKMEMCTLREG_MASK 0x1f /* only 5 bits used - only for Audigy */ - -/* Tank audio data is logarithmically compressed down to 16 bits before writing to TRAM and is */ -/* decompressed back to 20 bits on a read. There are a total of 160 locations, the last 32 */ -/* locations are for external TRAM. */ -#define TANKMEMDATAREGBASE 0x200 /* Tank memory data registers base */ -#define TANKMEMDATAREG_MASK 0x000fffff /* 20 bit tank audio data field */ - -/* Combined address field and memory opcode or flag field. 160 locations, last 32 are external */ -#define TANKMEMADDRREGBASE 0x300 /* Tank memory address registers base */ -#define TANKMEMADDRREG_ADDR_MASK 0x000fffff /* 20 bit tank address field */ -#define TANKMEMADDRREG_CLEAR 0x00800000 /* Clear tank memory */ -#define TANKMEMADDRREG_ALIGN 0x00400000 /* Align read or write relative to tank access */ -#define TANKMEMADDRREG_WRITE 0x00200000 /* Write to tank memory */ -#define TANKMEMADDRREG_READ 0x00100000 /* Read from tank memory */ -#define MICROCODEBASE 0x400 /* Microcode data base address */ +/* The public header defines the GPR and TRAM base addresses that + * are valid for _both_ CPU and DSP addressing. */ /* Each DSP microcode instruction is mapped into 2 doublewords */ /* NOTE: When writing, always write the LO doubleword first. Reads can be in either order. */ -#define LOWORD_OPX_MASK 0x000ffc00 /* Instruction operand X */ -#define LOWORD_OPY_MASK 0x000003ff /* Instruction operand Y */ -#define HIWORD_OPCODE_MASK 0x00f00000 /* Instruction opcode */ -#define HIWORD_RESULT_MASK 0x000ffc00 /* Instruction result */ -#define HIWORD_OPA_MASK 0x000003ff /* Instruction operand A */ +#define MICROCODEBASE 0x400 /* Microcode data base address */ +#define A_MICROCODEBASE 0x600 -/* Audigy Soundcard have a different instruction format */ -#define A_MICROCODEBASE 0x600 -#define A_LOWORD_OPY_MASK 0x000007ff -#define A_LOWORD_OPX_MASK 0x007ff000 -#define A_HIWORD_OPCODE_MASK 0x0f000000 -#define A_HIWORD_RESULT_MASK 0x007ff000 -#define A_HIWORD_OPA_MASK 0x000007ff +/************************************************************************************************/ +/* E-MU Digital Audio System overview */ +/************************************************************************************************/ + +// - These cards use a regular PCI-attached Audigy chip (Alice2/Tina/Tina2); +// the PCIe variants simply put the Audigy chip behind a PCI bridge. +// - All physical PCM I/O is routed through an additional FPGA; the regular +// EXTIN/EXTOUT ports are unconnected. +// - The FPGA has a signal routing matrix, to connect each destination (output +// socket or capture channel) to a source (input socket or playback channel). +// - The FPGA is controlled via Audigy's GPIO port, while sample data is +// transmitted via proprietary EMU32 serial links. On first-generation +// E-MU 1010 cards, Audigy's I2S inputs are also used for sample data. +// - The Audio/Micro Dock is attached to Hana via EDI, a "network" link. +// - The Audigy chip operates in slave mode; the clock is supplied by the FPGA. +// Gen1 E-MU 1010 cards have two crystals (for 44.1 kHz and 48 kHz multiples), +// while the later cards use a single crystal and a PLL chip. +// - The whole card is switched to 2x/4x mode to achieve 88.2/96/176.4/192 kHz +// sample rates. Alice2/Tina keeps running at 44.1/48 kHz, but multiple channels +// are bundled. +// - The number of available EMU32/EDI channels is hit in 2x/4x mode, so the total +// number of usable inputs/outputs is limited, esp. with ADAT in use. +// - S/PDIF is unavailable in 4x mode (only over TOSLINK on newer 1010 cards) due +// to being unspecified at 176.4/192 kHz. Therefore, the Dock's S/PDIF channels +// can overlap with the Dock's ADC/DAC's high channels. +// - The code names are mentioned below and in the emu_chip_details table. /************************************************************************************************/ -/* EMU1010m HANA FPGA registers */ +/* EMU1010 FPGA registers */ /************************************************************************************************/ + #define EMU_HANA_DESTHI 0x00 /* 0000xxx 3 bits Link Destination */ #define EMU_HANA_DESTLO 0x01 /* 00xxxxx 5 bits */ + #define EMU_HANA_SRCHI 0x02 /* 0000xxx 3 bits Link Source */ #define EMU_HANA_SRCLO 0x03 /* 00xxxxx 5 bits */ + #define EMU_HANA_DOCK_PWR 0x04 /* 000000x 1 bits Audio Dock power */ #define EMU_HANA_DOCK_PWR_ON 0x01 /* Audio Dock power on */ + #define EMU_HANA_WCLOCK 0x05 /* 0000xxx 3 bits Word Clock source select */ /* Must be written after power on to reset DLL */ /* One is unable to detect the Audio dock without this */ @@ -967,7 +965,7 @@ #define EMU_HANA_WCLOCK_INT_44_1K 0x01 #define EMU_HANA_WCLOCK_HANA_SPDIF_IN 0x02 #define EMU_HANA_WCLOCK_HANA_ADAT_IN 0x03 -#define EMU_HANA_WCLOCK_SYNC_BNCN 0x04 +#define EMU_HANA_WCLOCK_SYNC_BNC 0x04 #define EMU_HANA_WCLOCK_2ND_HANA 0x05 #define EMU_HANA_WCLOCK_SRC_RESERVED 0x06 #define EMU_HANA_WCLOCK_OFF 0x07 /* For testing, forces fallback to DEFCLOCK */ @@ -996,10 +994,10 @@ #define EMU_HANA_IRQ_DOCK_LOST 0x08 #define EMU_HANA_SPDIF_MODE 0x0a /* 00xxxxx 5 bits SPDIF MODE */ -#define EMU_HANA_SPDIF_MODE_TX_COMSUMER 0x00 +#define EMU_HANA_SPDIF_MODE_TX_CONSUMER 0x00 #define EMU_HANA_SPDIF_MODE_TX_PRO 0x01 #define EMU_HANA_SPDIF_MODE_TX_NOCOPY 0x02 -#define EMU_HANA_SPDIF_MODE_RX_COMSUMER 0x00 +#define EMU_HANA_SPDIF_MODE_RX_CONSUMER 0x00 #define EMU_HANA_SPDIF_MODE_RX_PRO 0x04 #define EMU_HANA_SPDIF_MODE_RX_NOCOPY 0x08 #define EMU_HANA_SPDIF_MODE_RX_INVALID 0x10 @@ -1011,8 +1009,12 @@ #define EMU_HANA_OPTICAL_OUT_ADAT 0x02 #define EMU_HANA_MIDI_IN 0x0c /* 000000x 1 bit Control MIDI */ -#define EMU_HANA_MIDI_IN_FROM_HAMOA 0x00 /* HAMOA MIDI in to Alice 2 MIDI B */ -#define EMU_HANA_MIDI_IN_FROM_DOCK 0x01 /* Audio Dock MIDI in to Alice 2 MIDI B */ +#define EMU_HANA_MIDI_INA_FROM_HAMOA 0x01 /* HAMOA MIDI in to Alice 2 MIDI A */ +#define EMU_HANA_MIDI_INA_FROM_DOCK1 0x02 /* Audio Dock-1 MIDI in to Alice 2 MIDI A */ +#define EMU_HANA_MIDI_INA_FROM_DOCK2 0x03 /* Audio Dock-2 MIDI in to Alice 2 MIDI A */ +#define EMU_HANA_MIDI_INB_FROM_HAMOA 0x08 /* HAMOA MIDI in to Alice 2 MIDI B */ +#define EMU_HANA_MIDI_INB_FROM_DOCK1 0x10 /* Audio Dock-1 MIDI in to Alice 2 MIDI B */ +#define EMU_HANA_MIDI_INB_FROM_DOCK2 0x18 /* Audio Dock-2 MIDI in to Alice 2 MIDI B */ #define EMU_HANA_DOCK_LEDS_1 0x0d /* 000xxxx 4 bit Audio Dock LEDs */ #define EMU_HANA_DOCK_LEDS_1_MIDI1 0x01 /* MIDI 1 LED on */ @@ -1037,51 +1039,49 @@ #define EMU_HANA_DOCK_LEDS_3_MANUAL_SIGNAL 0x20 /* Manual Signal detection */ #define EMU_HANA_ADC_PADS 0x10 /* 0000xxx 3 bit Audio Dock ADC 14dB pads */ -#define EMU_HANA_DOCK_ADC_PAD1 0x01 /* 14dB Attenuation on Audio Dock ADC 1 */ -#define EMU_HANA_DOCK_ADC_PAD2 0x02 /* 14dB Attenuation on Audio Dock ADC 2 */ -#define EMU_HANA_DOCK_ADC_PAD3 0x04 /* 14dB Attenuation on Audio Dock ADC 3 */ -#define EMU_HANA_0202_ADC_PAD1 0x08 /* 14dB Attenuation on 0202 ADC 1 */ +#define EMU_HANA_DOCK_ADC_PAD1 0x01 /* 14dB Attenuation on Audio Dock ADC 1 */ +#define EMU_HANA_DOCK_ADC_PAD2 0x02 /* 14dB Attenuation on Audio Dock ADC 2 */ +#define EMU_HANA_DOCK_ADC_PAD3 0x04 /* 14dB Attenuation on Audio Dock ADC 3 */ +#define EMU_HANA_0202_ADC_PAD1 0x08 /* 14dB Attenuation on 0202 ADC 1 */ #define EMU_HANA_DOCK_MISC 0x11 /* 0xxxxxx 6 bit Audio Dock misc bits */ -#define EMU_HANA_DOCK_DAC1_MUTE 0x01 /* DAC 1 Mute */ -#define EMU_HANA_DOCK_DAC2_MUTE 0x02 /* DAC 2 Mute */ -#define EMU_HANA_DOCK_DAC3_MUTE 0x04 /* DAC 3 Mute */ -#define EMU_HANA_DOCK_DAC4_MUTE 0x08 /* DAC 4 Mute */ +#define EMU_HANA_DOCK_DAC1_MUTE 0x01 /* DAC 1 Mute */ +#define EMU_HANA_DOCK_DAC2_MUTE 0x02 /* DAC 2 Mute */ +#define EMU_HANA_DOCK_DAC3_MUTE 0x04 /* DAC 3 Mute */ +#define EMU_HANA_DOCK_DAC4_MUTE 0x08 /* DAC 4 Mute */ #define EMU_HANA_DOCK_PHONES_192_DAC1 0x00 /* DAC 1 Headphones source at 192kHz */ #define EMU_HANA_DOCK_PHONES_192_DAC2 0x10 /* DAC 2 Headphones source at 192kHz */ #define EMU_HANA_DOCK_PHONES_192_DAC3 0x20 /* DAC 3 Headphones source at 192kHz */ #define EMU_HANA_DOCK_PHONES_192_DAC4 0x30 /* DAC 4 Headphones source at 192kHz */ #define EMU_HANA_MIDI_OUT 0x12 /* 00xxxxx 5 bit Source for each MIDI out port */ -#define EMU_HANA_MIDI_OUT_0202 0x01 /* 0202 MIDI from Alice 2. 0 = A, 1 = B */ -#define EMU_HANA_MIDI_OUT_DOCK1 0x02 /* Audio Dock MIDI1 front, from Alice 2. 0 = A, 1 = B */ -#define EMU_HANA_MIDI_OUT_DOCK2 0x04 /* Audio Dock MIDI2 rear, from Alice 2. 0 = A, 1 = B */ -#define EMU_HANA_MIDI_OUT_SYNC2 0x08 /* Sync card. Not the actual MIDI out jack. 0 = A, 1 = B */ -#define EMU_HANA_MIDI_OUT_LOOP 0x10 /* 0 = bits (3:0) normal. 1 = MIDI loopback enabled. */ +#define EMU_HANA_MIDI_OUT_0202 0x01 /* 0202 MIDI from Alice 2. 0 = A, 1 = B */ +#define EMU_HANA_MIDI_OUT_DOCK1 0x02 /* Audio Dock MIDI1 front, from Alice 2. 0 = A, 1 = B */ +#define EMU_HANA_MIDI_OUT_DOCK2 0x04 /* Audio Dock MIDI2 rear, from Alice 2. 0 = A, 1 = B */ +#define EMU_HANA_MIDI_OUT_SYNC2 0x08 /* Sync card. Not the actual MIDI out jack. 0 = A, 1 = B */ +#define EMU_HANA_MIDI_OUT_LOOP 0x10 /* 0 = bits (3:0) normal. 1 = MIDI loopback enabled. */ #define EMU_HANA_DAC_PADS 0x13 /* 00xxxxx 5 bit DAC 14dB attenuation pads */ -#define EMU_HANA_DOCK_DAC_PAD1 0x01 /* 14dB Attenuation on AudioDock DAC 1. Left and Right */ -#define EMU_HANA_DOCK_DAC_PAD2 0x02 /* 14dB Attenuation on AudioDock DAC 2. Left and Right */ -#define EMU_HANA_DOCK_DAC_PAD3 0x04 /* 14dB Attenuation on AudioDock DAC 3. Left and Right */ -#define EMU_HANA_DOCK_DAC_PAD4 0x08 /* 14dB Attenuation on AudioDock DAC 4. Left and Right */ -#define EMU_HANA_0202_DAC_PAD1 0x10 /* 14dB Attenuation on 0202 DAC 1. Left and Right */ +#define EMU_HANA_DOCK_DAC_PAD1 0x01 /* 14dB Attenuation on AudioDock DAC 1. Left and Right */ +#define EMU_HANA_DOCK_DAC_PAD2 0x02 /* 14dB Attenuation on AudioDock DAC 2. Left and Right */ +#define EMU_HANA_DOCK_DAC_PAD3 0x04 /* 14dB Attenuation on AudioDock DAC 3. Left and Right */ +#define EMU_HANA_DOCK_DAC_PAD4 0x08 /* 14dB Attenuation on AudioDock DAC 4. Left and Right */ +#define EMU_HANA_0202_DAC_PAD1 0x10 /* 14dB Attenuation on 0202 DAC 1. Left and Right */ /* 0x14 - 0x1f Unused R/W registers */ -#define EMU_HANA_IRQ_STATUS 0x20 /* 000xxxx 4 bits IRQ Status */ -#if 0 /* Already defined for reg 0x09 IRQ_ENABLE */ -#define EMU_HANA_IRQ_WCLK_CHANGED 0x01 -#define EMU_HANA_IRQ_ADAT 0x02 -#define EMU_HANA_IRQ_DOCK 0x04 -#define EMU_HANA_IRQ_DOCK_LOST 0x08 -#endif + +#define EMU_HANA_IRQ_STATUS 0x20 /* 00xxxxx 5 bits IRQ Status */ + /* Same bits as for EMU_HANA_IRQ_ENABLE */ + /* Reading the register resets it. */ #define EMU_HANA_OPTION_CARDS 0x21 /* 000xxxx 4 bits Presence of option cards */ -#define EMU_HANA_OPTION_HAMOA 0x01 /* HAMOA card present */ -#define EMU_HANA_OPTION_SYNC 0x02 /* Sync card present */ -#define EMU_HANA_OPTION_DOCK_ONLINE 0x04 /* Audio Dock online and FPGA configured */ -#define EMU_HANA_OPTION_DOCK_OFFLINE 0x08 /* Audio Dock online and FPGA not configured */ +#define EMU_HANA_OPTION_HAMOA 0x01 /* Hamoa (analog I/O) card present */ +#define EMU_HANA_OPTION_SYNC 0x02 /* Sync card present */ +#define EMU_HANA_OPTION_DOCK_ONLINE 0x04 /* Audio/Micro dock present and FPGA configured */ +#define EMU_HANA_OPTION_DOCK_OFFLINE 0x08 /* Audio/Micro dock present and FPGA not configured */ -#define EMU_HANA_ID 0x22 /* 1010101 7 bits ID byte & 0x7f = 0x55 */ +#define EMU_HANA_ID 0x22 /* 1010101 7 bits ID byte & 0x7f = 0x55 with Alice2 */ + /* 0010101 5 bits ID byte & 0x1f = 0x15 with Tina/2 */ #define EMU_HANA_MAJOR_REV 0x23 /* 0000xxx 3 bit Hana FPGA Major rev */ #define EMU_HANA_MINOR_REV 0x24 /* 0000xxx 3 bit Hana FPGA Minor rev */ @@ -1090,8 +1090,8 @@ #define EMU_DOCK_MINOR_REV 0x26 /* 0000xxx 3 bit Audio Dock FPGA Minor rev */ #define EMU_DOCK_BOARD_ID 0x27 /* 00000xx 2 bits Audio Dock ID pins */ -#define EMU_DOCK_BOARD_ID0 0x00 /* ID bit 0 */ -#define EMU_DOCK_BOARD_ID1 0x03 /* ID bit 1 */ +#define EMU_DOCK_BOARD_ID0 0x00 /* ID bit 0 */ +#define EMU_DOCK_BOARD_ID1 0x03 /* ID bit 1 */ #define EMU_HANA_WC_SPDIF_HI 0x28 /* 0xxxxxx 6 bit SPDIF IN Word clock, upper 6 bits */ #define EMU_HANA_WC_SPDIF_LO 0x29 /* 0xxxxxx 6 bit SPDIF IN Word clock, lower 6 bits */ @@ -1104,31 +1104,35 @@ #define EMU_HANA2_WC_SPDIF_HI 0x2e /* 0xxxxxx 6 bit HANA2 SPDIF IN Word clock, upper 6 bits */ #define EMU_HANA2_WC_SPDIF_LO 0x2f /* 0xxxxxx 6 bit HANA2 SPDIF IN Word clock, lower 6 bits */ + /* 0x30 - 0x3f Unused Read only registers */ +// The meaning of this is not clear; kX-project just calls it "lock" in some info-only code. +#define EMU_HANA_LOCK_STS_LO 0x38 /* 0xxxxxx lower 6 bits */ +#define EMU_HANA_LOCK_STS_HI 0x39 /* 0xxxxxx upper 6 bits */ + /************************************************************************************************/ -/* EMU1010m HANA Destinations */ +/* EMU1010 Audio Destinations */ /************************************************************************************************/ -/* Hana, original 1010,1212,1820 using Alice2 - * Destiniations for SRATEX = 1X rates: 44.1 kHz or 48 kHz +/* Hana, original 1010,1212m,1820[m] using Alice2 * 0x00, 0x00-0x0f: 16 EMU32 channels to Alice2 - * 0x01, 0x10-0x1f: 32 Elink channels to Audio Dock - * 0x01, 0x00: Dock DAC 1 Left - * 0x01, 0x04: Dock DAC 1 Right - * 0x01, 0x08: Dock DAC 2 Left - * 0x01, 0x0c: Dock DAC 2 Right - * 0x01, 0x10: Dock DAC 3 Left - * 0x01, 0x12: PHONES Left - * 0x01, 0x14: Dock DAC 3 Right - * 0x01, 0x16: PHONES Right - * 0x01, 0x18: Dock DAC 4 Left - * 0x01, 0x1a: S/PDIF Left - * 0x01, 0x1c: Dock DAC 4 Right - * 0x01, 0x1e: S/PDIF Right + * 0x01, 0x00-0x1f: 32 EDI channels to Audio Dock + * 0x00: Dock DAC 1 Left + * 0x04: Dock DAC 1 Right + * 0x08: Dock DAC 2 Left + * 0x0c: Dock DAC 2 Right + * 0x10: Dock DAC 3 Left + * 0x12: PHONES Left (n/a in 2x/4x mode; output mirrors DAC4 Left) + * 0x14: Dock DAC 3 Right + * 0x16: PHONES Right (n/a in 2x/4x mode; output mirrors DAC4 Right) + * 0x18: Dock DAC 4 Left + * 0x1a: S/PDIF Left + * 0x1c: Dock DAC 4 Right + * 0x1e: S/PDIF Right * 0x02, 0x00: Hana S/PDIF Left * 0x02, 0x01: Hana S/PDIF Right - * 0x03, 0x00: Hanoa DAC Left - * 0x03, 0x01: Hanoa DAC Right + * 0x03, 0x00: Hamoa DAC Left + * 0x03, 0x01: Hamoa DAC Right * 0x04, 0x00-0x07: Hana ADAT * 0x05, 0x00: I2S0 Left to Alice2 * 0x05, 0x01: I2S0 Right to Alice2 @@ -1140,40 +1144,29 @@ * Hana2 never released, but used Tina * Not needed. * - * Hana3, rev2 1010,1212,1616 using Tina - * Destinations for SRATEX = 1X rates: 44.1 kHz or 48 kHz + * Hana3, rev2 1010,1212m,1616[m] using Tina * 0x00, 0x00-0x0f: 16 EMU32A channels to Tina - * 0x01, 0x10-0x1f: 32 EDI channels to Micro Dock - * 0x01, 0x00: Dock DAC 1 Left - * 0x01, 0x04: Dock DAC 1 Right - * 0x01, 0x08: Dock DAC 2 Left - * 0x01, 0x0c: Dock DAC 2 Right - * 0x01, 0x10: Dock DAC 3 Left - * 0x01, 0x12: Dock S/PDIF Left - * 0x01, 0x14: Dock DAC 3 Right - * 0x01, 0x16: Dock S/PDIF Right - * 0x01, 0x18-0x1f: Dock ADAT 0-7 + * 0x01, 0x00-0x1f: 32 EDI channels to Micro Dock + * 0x00: Dock DAC 1 Left + * 0x04: Dock DAC 1 Right + * 0x08: Dock DAC 2 Left + * 0x0c: Dock DAC 2 Right + * 0x10: Dock DAC 3 Left + * 0x12: Dock S/PDIF Left + * 0x14: Dock DAC 3 Right + * 0x16: Dock S/PDIF Right + * 0x18-0x1f: Dock ADAT 0-7 * 0x02, 0x00: Hana3 S/PDIF Left * 0x02, 0x01: Hana3 S/PDIF Right - * 0x03, 0x00: Hanoa DAC Left - * 0x03, 0x01: Hanoa DAC Right + * 0x03, 0x00: Hamoa DAC Left + * 0x03, 0x01: Hamoa DAC Right * 0x04, 0x00-0x07: Hana3 ADAT 0-7 * 0x05, 0x00-0x0f: 16 EMU32B channels to Tina * 0x06-0x07: Not used * * HanaLite, rev1 0404 using Alice2 - * Destiniations for SRATEX = 1X rates: 44.1 kHz or 48 kHz - * 0x00, 0x00-0x0f: 16 EMU32 channels to Alice2 - * 0x01: Not used - * 0x02, 0x00: S/PDIF Left - * 0x02, 0x01: S/PDIF Right - * 0x03, 0x00: DAC Left - * 0x03, 0x01: DAC Right - * 0x04-0x07: Not used - * - * HanaLiteLite, rev2 0404 using Alice2 - * Destiniations for SRATEX = 1X rates: 44.1 kHz or 48 kHz - * 0x00, 0x00-0x0f: 16 EMU32 channels to Alice2 + * HanaLiteLite, rev2 0404 using Tina + * 0x00, 0x00-0x0f: 16 EMU32 channels to Alice2/Tina * 0x01: Not used * 0x02, 0x00: S/PDIF Left * 0x02, 0x01: S/PDIF Right @@ -1182,35 +1175,21 @@ * 0x04-0x07: Not used * * Mana, Cardbus 1616 using Tina2 - * Destinations for SRATEX = 1X rates: 44.1 kHz or 48 kHz * 0x00, 0x00-0x0f: 16 EMU32A channels to Tina2 - * 0x01, 0x10-0x1f: 32 EDI channels to Micro Dock - * 0x01, 0x00: Dock DAC 1 Left - * 0x01, 0x04: Dock DAC 1 Right - * 0x01, 0x08: Dock DAC 2 Left - * 0x01, 0x0c: Dock DAC 2 Right - * 0x01, 0x10: Dock DAC 3 Left - * 0x01, 0x12: Dock S/PDIF Left - * 0x01, 0x14: Dock DAC 3 Right - * 0x01, 0x16: Dock S/PDIF Right - * 0x01, 0x18-0x1f: Dock ADAT 0-7 + * 0x01, 0x00-0x1f: 32 EDI channels to Micro Dock + * (same as rev2 1010) * 0x02: Not used * 0x03, 0x00: Mana DAC Left * 0x03, 0x01: Mana DAC Right * 0x04, 0x00-0x0f: 16 EMU32B channels to Tina2 * 0x05-0x07: Not used - * - * */ + /* 32-bit destinations of signal in the Hana FPGA. Destinations are either - * physical outputs of Hana, or outputs going to Alice2 (audigy) for capture - * - 16 x EMU_DST_ALICE2_EMU32_X. - */ -/* EMU32 = 32-bit serial channel between Alice2 (audigy) and Hana (FPGA) */ -/* EMU_DST_ALICE2_EMU32_X - data channels from Hana to Alice2 used for capture. - * Which data is fed into a EMU_DST_ALICE2_EMU32_X channel in Hana depends on - * setup of mixer control for each destination - see emumixer.c - - * snd_emu1010_output_enum_ctls[], snd_emu1010_input_enum_ctls[] + * physical outputs of Hana, or outputs going to Alice2/Tina for capture - + * 16 x EMU_DST_ALICE2_EMU32_X (2x on rev2 boards). Which data is fed into + * a channel depends on the mixer control setting for each destination - see + * emumixer.c - snd_emu1010_output_enum_ctls[], snd_emu1010_input_enum_ctls[] */ #define EMU_DST_ALICE2_EMU32_0 0x000f /* 16 EMU32 channels to Alice2 +0 to +0xf */ #define EMU_DST_ALICE2_EMU32_1 0x0000 /* 16 EMU32 channels to Alice2 +0 to +0xf */ @@ -1270,8 +1249,12 @@ #define EMU_DST_DOCK_SPDIF_RIGHT2 0x011f /* Audio Dock SPDIF Right, 2nd or 96kHz */ #define EMU_DST_HANA_SPDIF_LEFT1 0x0200 /* Hana SPDIF Left, 1st or 48kHz only */ #define EMU_DST_HANA_SPDIF_LEFT2 0x0202 /* Hana SPDIF Left, 2nd or 96kHz */ +#define EMU_DST_HANA_SPDIF_LEFT3 0x0204 /* Hana SPDIF Left, 3rd or 192kHz */ +#define EMU_DST_HANA_SPDIF_LEFT4 0x0206 /* Hana SPDIF Left, 4th or 192kHz */ #define EMU_DST_HANA_SPDIF_RIGHT1 0x0201 /* Hana SPDIF Right, 1st or 48kHz only */ #define EMU_DST_HANA_SPDIF_RIGHT2 0x0203 /* Hana SPDIF Right, 2nd or 96kHz */ +#define EMU_DST_HANA_SPDIF_RIGHT3 0x0205 /* Hana SPDIF Right, 3rd or 192kHz */ +#define EMU_DST_HANA_SPDIF_RIGHT4 0x0207 /* Hana SPDIF Right, 4th or 192kHz */ #define EMU_DST_HAMOA_DAC_LEFT1 0x0300 /* Hamoa DAC Left, 1st or 48kHz only */ #define EMU_DST_HAMOA_DAC_LEFT2 0x0302 /* Hamoa DAC Left, 2nd or 96kHz */ #define EMU_DST_HAMOA_DAC_LEFT3 0x0304 /* Hamoa DAC Left, 3rd or 192kHz */ @@ -1280,6 +1263,7 @@ #define EMU_DST_HAMOA_DAC_RIGHT2 0x0303 /* Hamoa DAC Right, 2nd or 96kHz */ #define EMU_DST_HAMOA_DAC_RIGHT3 0x0305 /* Hamoa DAC Right, 3rd or 192kHz */ #define EMU_DST_HAMOA_DAC_RIGHT4 0x0307 /* Hamoa DAC Right, 4th or 192kHz */ +// In S/MUX mode, the samples of one channel are adjacent. #define EMU_DST_HANA_ADAT 0x0400 /* Hana ADAT 8 channel out +0 to +7 */ #define EMU_DST_ALICE_I2S0_LEFT 0x0500 /* Alice2 I2S0 Left */ #define EMU_DST_ALICE_I2S0_RIGHT 0x0501 /* Alice2 I2S0 Right */ @@ -1289,39 +1273,32 @@ #define EMU_DST_ALICE_I2S2_RIGHT 0x0701 /* Alice2 I2S2 Right */ /* Additional destinations for 1616(M)/Microdock */ -/* Microdock S/PDIF OUT Left, 1st or 48kHz only */ -#define EMU_DST_MDOCK_SPDIF_LEFT1 0x0112 -/* Microdock S/PDIF OUT Left, 2nd or 96kHz */ -#define EMU_DST_MDOCK_SPDIF_LEFT2 0x0113 -/* Microdock S/PDIF OUT Right, 1st or 48kHz only */ -#define EMU_DST_MDOCK_SPDIF_RIGHT1 0x0116 -/* Microdock S/PDIF OUT Right, 2nd or 96kHz */ -#define EMU_DST_MDOCK_SPDIF_RIGHT2 0x0117 -/* Microdock S/PDIF ADAT 8 channel out +8 to +f */ -#define EMU_DST_MDOCK_ADAT 0x0118 - -/* Headphone jack on 1010 cardbus? 44.1/48kHz only? */ -#define EMU_DST_MANA_DAC_LEFT 0x0300 -/* Headphone jack on 1010 cardbus? 44.1/48kHz only? */ -#define EMU_DST_MANA_DAC_RIGHT 0x0301 + +#define EMU_DST_MDOCK_SPDIF_LEFT1 0x0112 /* Microdock S/PDIF OUT Left, 1st or 48kHz only */ +#define EMU_DST_MDOCK_SPDIF_LEFT2 0x0113 /* Microdock S/PDIF OUT Left, 2nd or 96kHz */ +#define EMU_DST_MDOCK_SPDIF_RIGHT1 0x0116 /* Microdock S/PDIF OUT Right, 1st or 48kHz only */ +#define EMU_DST_MDOCK_SPDIF_RIGHT2 0x0117 /* Microdock S/PDIF OUT Right, 2nd or 96kHz */ +#define EMU_DST_MDOCK_ADAT 0x0118 /* Microdock S/PDIF ADAT 8 channel out +8 to +f */ + +#define EMU_DST_MANA_DAC_LEFT 0x0300 /* Headphone jack on 1010 cardbus? 44.1/48kHz only? */ +#define EMU_DST_MANA_DAC_RIGHT 0x0301 /* Headphone jack on 1010 cardbus? 44.1/48kHz only? */ /************************************************************************************************/ -/* EMU1010m HANA Sources */ +/* EMU1010 Audio Sources */ /************************************************************************************************/ -/* Hana, original 1010,1212,1820 using Alice2 - * Sources SRATEX = 1X rates: 44.1 kHz or 48 kHz - * 0x00,0x00-0x1f: Silence - * 0x01, 0x10-0x1f: 32 Elink channels from Audio Dock - * 0x01, 0x00: Dock Mic A - * 0x01, 0x04: Dock Mic B - * 0x01, 0x08: Dock ADC 1 Left - * 0x01, 0x0c: Dock ADC 1 Right - * 0x01, 0x10: Dock ADC 2 Left - * 0x01, 0x14: Dock ADC 2 Right - * 0x01, 0x18: Dock ADC 3 Left - * 0x01, 0x1c: Dock ADC 3 Right - * 0x02, 0x00: Hana ADC Left - * 0x02, 0x01: Hana ADC Right +/* Hana, original 1010,1212m,1820[m] using Alice2 + * 0x00, 0x00-0x1f: Silence + * 0x01, 0x00-0x1f: 32 EDI channels from Audio Dock + * 0x00: Dock Mic A + * 0x04: Dock Mic B + * 0x08: Dock ADC 1 Left + * 0x0c: Dock ADC 1 Right + * 0x10: Dock ADC 2 Left + * 0x14: Dock ADC 2 Right + * 0x18: Dock ADC 3 Left + * 0x1c: Dock ADC 3 Right + * 0x02, 0x00: Hamoa ADC Left + * 0x02, 0x01: Hamoa ADC Right * 0x03, 0x00-0x0f: 16 inputs from Alice2 Emu32A output * 0x03, 0x10-0x1f: 16 inputs from Alice2 Emu32B output * 0x04, 0x00-0x07: Hana ADAT @@ -1332,23 +1309,20 @@ * Hana2 never released, but used Tina * Not needed. * - * Hana3, rev2 1010,1212,1616 using Tina - * Sources SRATEX = 1X rates: 44.1 kHz or 48 kHz - * 0x00,0x00-0x1f: Silence - * 0x01, 0x10-0x1f: 32 Elink channels from Audio Dock - * 0x01, 0x00: Dock Mic A - * 0x01, 0x04: Dock Mic B - * 0x01, 0x08: Dock ADC 1 Left - * 0x01, 0x0c: Dock ADC 1 Right - * 0x01, 0x10: Dock ADC 2 Left - * 0x01, 0x12: Dock S/PDIF Left - * 0x01, 0x14: Dock ADC 2 Right - * 0x01, 0x16: Dock S/PDIF Right - * 0x01, 0x18-0x1f: Dock ADAT 0-7 - * 0x01, 0x18: Dock ADC 3 Left - * 0x01, 0x1c: Dock ADC 3 Right - * 0x02, 0x00: Hanoa ADC Left - * 0x02, 0x01: Hanoa ADC Right + * Hana3, rev2 1010,1212m,1616[m] using Tina + * 0x00, 0x00-0x1f: Silence + * 0x01, 0x00-0x1f: 32 EDI channels from Micro Dock + * 0x00: Dock Mic A + * 0x04: Dock Mic B + * 0x08: Dock ADC 1 Left + * 0x0c: Dock ADC 1 Right + * 0x10: Dock ADC 2 Left + * 0x12: Dock S/PDIF Left + * 0x14: Dock ADC 2 Right + * 0x16: Dock S/PDIF Right + * 0x18-0x1f: Dock ADAT 0-7 + * 0x02, 0x00: Hamoa ADC Left + * 0x02, 0x01: Hamoa ADC Right * 0x03, 0x00-0x0f: 16 inputs from Tina Emu32A output * 0x03, 0x10-0x1f: 16 inputs from Tina Emu32B output * 0x04, 0x00-0x07: Hana3 ADAT @@ -1357,58 +1331,32 @@ * 0x06-0x07: Not used * * HanaLite, rev1 0404 using Alice2 - * Sources SRATEX = 1X rates: 44.1 kHz or 48 kHz - * 0x00,0x00-0x1f: Silence - * 0x01: Not used - * 0x02, 0x00: ADC Left - * 0x02, 0x01: ADC Right - * 0x03, 0x00-0x0f: 16 inputs from Alice2 Emu32A output - * 0x03, 0x10-0x1f: 16 inputs from Alice2 Emu32B output - * 0x04: Not used - * 0x05, 0x00: S/PDIF Left - * 0x05, 0x01: S/PDIF Right - * 0x06-0x07: Not used - * - * HanaLiteLite, rev2 0404 using Alice2 - * Sources SRATEX = 1X rates: 44.1 kHz or 48 kHz - * 0x00,0x00-0x1f: Silence + * HanaLiteLite, rev2 0404 using Tina + * 0x00, 0x00-0x1f: Silence * 0x01: Not used * 0x02, 0x00: ADC Left * 0x02, 0x01: ADC Right - * 0x03, 0x00-0x0f: 16 inputs from Alice2 Emu32A output - * 0x03, 0x10-0x1f: 16 inputs from Alice2 Emu32B output + * 0x03, 0x00-0x0f: 16 inputs from Alice2/Tina Emu32A output + * 0x03, 0x10-0x1f: 16 inputs from Alice2/Tina Emu32B output * 0x04: Not used * 0x05, 0x00: S/PDIF Left * 0x05, 0x01: S/PDIF Right * 0x06-0x07: Not used * * Mana, Cardbus 1616 using Tina2 - * Sources SRATEX = 1X rates: 44.1 kHz or 48 kHz - * 0x00,0x00-0x1f: Silence - * 0x01, 0x10-0x1f: 32 Elink channels from Audio Dock - * 0x01, 0x00: Dock Mic A - * 0x01, 0x04: Dock Mic B - * 0x01, 0x08: Dock ADC 1 Left - * 0x01, 0x0c: Dock ADC 1 Right - * 0x01, 0x10: Dock ADC 2 Left - * 0x01, 0x12: Dock S/PDIF Left - * 0x01, 0x14: Dock ADC 2 Right - * 0x01, 0x16: Dock S/PDIF Right - * 0x01, 0x18-0x1f: Dock ADAT 0-7 - * 0x01, 0x18: Dock ADC 3 Left - * 0x01, 0x1c: Dock ADC 3 Right + * 0x00, 0x00-0x1f: Silence + * 0x01, 0x00-0x1f: 32 EDI channels from Micro Dock + * (same as rev2 1010) * 0x02: Not used - * 0x03, 0x00-0x0f: 16 inputs from Tina Emu32A output - * 0x03, 0x10-0x1f: 16 inputs from Tina Emu32B output + * 0x03, 0x00-0x0f: 16 inputs from Tina2 Emu32A output + * 0x03, 0x10-0x1f: 16 inputs from Tina2 Emu32B output * 0x04-0x07: Not used - * */ /* 32-bit sources of signal in the Hana FPGA. The sources are routed to - * destinations using mixer control for each destination - see emumixer.c - * Sources are either physical inputs of FPGA, - * or outputs from Alice (audigy) - 16 x EMU_SRC_ALICE_EMU32A + - * 16 x EMU_SRC_ALICE_EMU32B + * destinations using a mixer control for each destination - see emumixer.c. + * Sources are either physical inputs of Hana, or inputs from Alice2/Tina - + * 16 x EMU_SRC_ALICE_EMU32A + 16 x EMU_SRC_ALICE_EMU32B. */ #define EMU_SRC_SILENCE 0x0000 /* Silence */ #define EMU_SRC_DOCK_MIC_A1 0x0100 /* Audio Dock Mic A, 1st or 48kHz only */ @@ -1453,23 +1401,24 @@ #define EMU_SRC_HAMOA_ADC_RIGHT4 0x0207 /* Hamoa ADC Right, 4th or 192kHz */ #define EMU_SRC_ALICE_EMU32A 0x0300 /* Alice2 EMU32a 16 outputs. +0 to +0xf */ #define EMU_SRC_ALICE_EMU32B 0x0310 /* Alice2 EMU32b 16 outputs. +0 to +0xf */ +// In S/MUX mode, the samples of one channel are adjacent. #define EMU_SRC_HANA_ADAT 0x0400 /* Hana ADAT 8 channel in +0 to +7 */ #define EMU_SRC_HANA_SPDIF_LEFT1 0x0500 /* Hana SPDIF Left, 1st or 48kHz only */ #define EMU_SRC_HANA_SPDIF_LEFT2 0x0502 /* Hana SPDIF Left, 2nd or 96kHz */ +#define EMU_SRC_HANA_SPDIF_LEFT3 0x0504 /* Hana SPDIF Left, 3rd or 192kHz */ +#define EMU_SRC_HANA_SPDIF_LEFT4 0x0506 /* Hana SPDIF Left, 4th or 192kHz */ #define EMU_SRC_HANA_SPDIF_RIGHT1 0x0501 /* Hana SPDIF Right, 1st or 48kHz only */ #define EMU_SRC_HANA_SPDIF_RIGHT2 0x0503 /* Hana SPDIF Right, 2nd or 96kHz */ +#define EMU_SRC_HANA_SPDIF_RIGHT3 0x0505 /* Hana SPDIF Right, 3rd or 192kHz */ +#define EMU_SRC_HANA_SPDIF_RIGHT4 0x0507 /* Hana SPDIF Right, 4th or 192kHz */ /* Additional inputs for 1616(M)/Microdock */ -/* Microdock S/PDIF Left, 1st or 48kHz only */ -#define EMU_SRC_MDOCK_SPDIF_LEFT1 0x0112 -/* Microdock S/PDIF Left, 2nd or 96kHz */ -#define EMU_SRC_MDOCK_SPDIF_LEFT2 0x0113 -/* Microdock S/PDIF Right, 1st or 48kHz only */ -#define EMU_SRC_MDOCK_SPDIF_RIGHT1 0x0116 -/* Microdock S/PDIF Right, 2nd or 96kHz */ -#define EMU_SRC_MDOCK_SPDIF_RIGHT2 0x0117 -/* Microdock ADAT 8 channel in +8 to +f */ -#define EMU_SRC_MDOCK_ADAT 0x0118 + +#define EMU_SRC_MDOCK_SPDIF_LEFT1 0x0112 /* Microdock S/PDIF Left, 1st or 48kHz only */ +#define EMU_SRC_MDOCK_SPDIF_LEFT2 0x0113 /* Microdock S/PDIF Left, 2nd or 96kHz */ +#define EMU_SRC_MDOCK_SPDIF_RIGHT1 0x0116 /* Microdock S/PDIF Right, 1st or 48kHz only */ +#define EMU_SRC_MDOCK_SPDIF_RIGHT2 0x0117 /* Microdock S/PDIF Right, 2nd or 96kHz */ +#define EMU_SRC_MDOCK_ADAT 0x0118 /* Microdock ADAT 8 channel in +8 to +f */ /* 0x600 and 0x700 no used */ @@ -1485,7 +1434,6 @@ enum { struct snd_emu10k1; struct snd_emu10k1_voice { - struct snd_emu10k1 *emu; int number; unsigned int use: 1, pcm: 1, @@ -1599,10 +1547,8 @@ struct snd_emu10k1_fx8010_pcm { }; struct snd_emu10k1_fx8010 { - unsigned short fxbus_mask; /* used FX buses (bitmask) */ - unsigned short extin_mask; /* used external inputs (bitmask) */ - unsigned short extout_mask; /* used external outputs (bitmask) */ - unsigned short pad1; + unsigned short extin_mask; /* used external inputs (bitmask); not used for Audigy */ + unsigned short extout_mask; /* used external outputs (bitmask); not used for Audigy */ unsigned int itram_size; /* internal TRAM size in samples */ struct snd_dma_buffer etram_pages; /* external TRAM pages and size */ unsigned int dbg; /* FX debugger register */ @@ -1639,14 +1585,26 @@ enum { EMU_MODEL_EMU0404, }; +// Chip-o-logy: +// - All SB Live! cards use EMU10K1 chips +// - All SB Audigy cards use CA* chips, termed "emu10k2" by the driver +// - Original Audigy uses CA0100 "Alice" +// - Audigy 2 uses CA0102/CA10200 "Alice2" +// - Has an interface for CA0151 (P16V) "Alice3" +// - Audigy 2 Value uses CA0108/CA10300 "Tina" +// - Approximately a CA0102 with an on-chip CA0151 (P17V) +// - Audigy 2 ZS NB uses CA0109 "Tina2" +// - Cardbus version of CA0108 struct snd_emu_chip_details { u32 vendor; u32 device; u32 subsystem; unsigned char revision; unsigned char emu10k1_chip; /* Original SB Live. Not SB Live 24bit. */ + /* Redundant with emu10k2_chip being unset. */ unsigned char emu10k2_chip; /* Audigy 1 or Audigy 2. */ unsigned char ca0102_chip; /* Audigy 1 or Audigy 2. Not SB Audigy 2 Value. */ + /* Redundant with ca0108_chip being unset. */ unsigned char ca0108_chip; /* Audigy 2 Value */ unsigned char ca_cardbus_chip; /* Audigy 2 ZS Notebook */ unsigned char ca0151_chip; /* P16V */ @@ -1656,8 +1614,8 @@ struct snd_emu_chip_details { unsigned char ac97_chip; /* Has an AC97 chip: 1 = mandatory, 2 = optional */ unsigned char ecard; /* APS EEPROM */ unsigned char emu_model; /* EMU model type */ - unsigned char spi_dac; /* SPI interface for DAC */ - unsigned char i2c_adc; /* I2C interface for ADC */ + unsigned char spi_dac; /* SPI interface for DAC; requires ca0108_chip */ + unsigned char i2c_adc; /* I2C interface for ADC; requires ca0108_chip */ unsigned char adc_1361t; /* Use Philips 1361T ADC */ unsigned char invert_shared_spdif; /* analog/digital switch inverted */ const char *driver; @@ -1691,7 +1649,6 @@ struct snd_emu10k1 { unsigned int revision; /* chip revision */ unsigned int serial; /* serial number */ unsigned short model; /* subsystem id */ - unsigned int card_type; /* EMU10K1_CARD_* */ unsigned int ecard_ctrl; /* ecard control bits */ unsigned int address_mode; /* address mode */ unsigned long dma_mask; /* PCI DMA mask */ @@ -1732,15 +1689,13 @@ struct snd_emu10k1 { void *synth; int (*get_synth_voice)(struct snd_emu10k1 *emu); - spinlock_t reg_lock; - spinlock_t emu_lock; - spinlock_t voice_lock; + spinlock_t reg_lock; // high-level driver lock + spinlock_t emu_lock; // low-level i/o lock + spinlock_t voice_lock; // voice allocator lock spinlock_t spi_lock; /* serialises access to spi port */ spinlock_t i2c_lock; /* serialises access to i2c port */ struct snd_emu10k1_voice voices[NUM_G]; - struct snd_emu10k1_voice p16v_voices[4]; - struct snd_emu10k1_voice p16v_capture_voice; int p16v_device_offset; u32 p16v_capture_source; u32 p16v_capture_channel; @@ -1760,11 +1715,11 @@ struct snd_emu10k1 { void (*capture_efx_interrupt)(struct snd_emu10k1 *emu, unsigned int status); void (*spdif_interrupt)(struct snd_emu10k1 *emu, unsigned int status); void (*dsp_interrupt)(struct snd_emu10k1 *emu); + void (*p16v_interrupt)(struct snd_emu10k1 *emu); struct snd_pcm_substream *pcm_capture_substream; struct snd_pcm_substream *pcm_capture_mic_substream; struct snd_pcm_substream *pcm_capture_efx_substream; - struct snd_pcm_substream *pcm_playback_efx_substream; struct snd_timer *timer; @@ -1824,9 +1779,9 @@ unsigned int snd_emu10k1_ptr20_read(struct snd_emu10k1 * emu, unsigned int reg, void snd_emu10k1_ptr20_write(struct snd_emu10k1 *emu, unsigned int reg, unsigned int chn, unsigned int data); int snd_emu10k1_spi_write(struct snd_emu10k1 * emu, unsigned int data); int snd_emu10k1_i2c_write(struct snd_emu10k1 *emu, u32 reg, u32 value); -int snd_emu1010_fpga_write(struct snd_emu10k1 * emu, u32 reg, u32 value); -int snd_emu1010_fpga_read(struct snd_emu10k1 * emu, u32 reg, u32 *value); -int snd_emu1010_fpga_link_dst_src_write(struct snd_emu10k1 * emu, u32 dst, u32 src); +void snd_emu1010_fpga_write(struct snd_emu10k1 *emu, u32 reg, u32 value); +void snd_emu1010_fpga_read(struct snd_emu10k1 *emu, u32 reg, u32 *value); +void snd_emu1010_fpga_link_dst_src_write(struct snd_emu10k1 *emu, u32 dst, u32 src); unsigned int snd_emu10k1_efx_read(struct snd_emu10k1 *emu, unsigned int pc); void snd_emu10k1_intr_enable(struct snd_emu10k1 *emu, unsigned int intrenb); void snd_emu10k1_intr_disable(struct snd_emu10k1 *emu, unsigned int intrenb); diff --git a/include/sound/pcm-indirect.h b/include/sound/pcm-indirect.h index 04127686e8d0..98e06ea73b2b 100644 --- a/include/sound/pcm-indirect.h +++ b/include/sound/pcm-indirect.h @@ -44,7 +44,7 @@ snd_pcm_indirect_playback_transfer(struct snd_pcm_substream *substream, if (diff < -(snd_pcm_sframes_t) (runtime->boundary / 2)) diff += runtime->boundary; if (diff < 0) - return -EINVAL; + return -EPIPE; rec->sw_ready += (int)frames_to_bytes(runtime, diff); rec->appl_ptr = appl_ptr; } @@ -83,6 +83,8 @@ snd_pcm_indirect_playback_pointer(struct snd_pcm_substream *substream, struct snd_pcm_indirect *rec, unsigned int ptr) { int bytes = ptr - rec->hw_io; + int err; + if (bytes < 0) bytes += rec->hw_buffer_size; rec->hw_io = ptr; @@ -90,8 +92,11 @@ snd_pcm_indirect_playback_pointer(struct snd_pcm_substream *substream, rec->sw_io += bytes; if (rec->sw_io >= rec->sw_buffer_size) rec->sw_io -= rec->sw_buffer_size; - if (substream->ops->ack) - substream->ops->ack(substream); + if (substream->ops->ack) { + err = substream->ops->ack(substream); + if (err == -EPIPE) + return SNDRV_PCM_POS_XRUN; + } return bytes_to_frames(substream->runtime, rec->sw_io); } @@ -112,7 +117,7 @@ snd_pcm_indirect_capture_transfer(struct snd_pcm_substream *substream, if (diff < -(snd_pcm_sframes_t) (runtime->boundary / 2)) diff += runtime->boundary; if (diff < 0) - return -EINVAL; + return -EPIPE; rec->sw_ready -= frames_to_bytes(runtime, diff); rec->appl_ptr = appl_ptr; } @@ -152,6 +157,8 @@ snd_pcm_indirect_capture_pointer(struct snd_pcm_substream *substream, { int qsize; int bytes = ptr - rec->hw_io; + int err; + if (bytes < 0) bytes += rec->hw_buffer_size; rec->hw_io = ptr; @@ -162,8 +169,11 @@ snd_pcm_indirect_capture_pointer(struct snd_pcm_substream *substream, rec->sw_io += bytes; if (rec->sw_io >= rec->sw_buffer_size) rec->sw_io -= rec->sw_buffer_size; - if (substream->ops->ack) - substream->ops->ack(substream); + if (substream->ops->ack) { + err = substream->ops->ack(substream); + if (err == -EPIPE) + return SNDRV_PCM_POS_XRUN; + } return bytes_to_frames(substream->runtime, rec->sw_io); } diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 27040b472a4f..19f564606ac4 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -378,18 +378,18 @@ struct snd_pcm_runtime { unsigned int rate_den; unsigned int no_period_wakeup: 1; - /* -- SW params -- */ - int tstamp_mode; /* mmap timestamp is updated */ + /* -- SW params; see struct snd_pcm_sw_params for comments -- */ + int tstamp_mode; unsigned int period_step; snd_pcm_uframes_t start_threshold; snd_pcm_uframes_t stop_threshold; - snd_pcm_uframes_t silence_threshold; /* Silence filling happens when - noise is nearest than this */ - snd_pcm_uframes_t silence_size; /* Silence filling size */ - snd_pcm_uframes_t boundary; /* pointers wrap point */ + snd_pcm_uframes_t silence_threshold; + snd_pcm_uframes_t silence_size; + snd_pcm_uframes_t boundary; + /* internal data of auto-silencer */ snd_pcm_uframes_t silence_start; /* starting pointer to silence area */ - snd_pcm_uframes_t silence_filled; /* size filled with silence */ + snd_pcm_uframes_t silence_filled; /* already filled part of silence area */ union snd_pcm_sync_id sync; /* hardware synchronization ID */ diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index de6810e94abe..0aa955aa8246 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -429,9 +429,14 @@ struct snd_pcm_sw_params { snd_pcm_uframes_t avail_min; /* min avail frames for wakeup */ snd_pcm_uframes_t xfer_align; /* obsolete: xfer size need to be a multiple */ snd_pcm_uframes_t start_threshold; /* min hw_avail frames for automatic start */ - snd_pcm_uframes_t stop_threshold; /* min avail frames for automatic stop */ - snd_pcm_uframes_t silence_threshold; /* min distance from noise for silence filling */ - snd_pcm_uframes_t silence_size; /* silence block size */ + /* + * The following two thresholds alleviate playback buffer underruns; when + * hw_avail drops below the threshold, the respective action is triggered: + */ + snd_pcm_uframes_t stop_threshold; /* - stop playback */ + snd_pcm_uframes_t silence_threshold; /* - pre-fill buffer with silence */ + snd_pcm_uframes_t silence_size; /* max size of silence pre-fill; when >= boundary, + * fill played area with silence immediately */ snd_pcm_uframes_t boundary; /* pointers wrap point */ unsigned int proto; /* protocol version */ unsigned int tstamp_type; /* timestamp type (req. proto >= 2.0.12) */ @@ -570,7 +575,8 @@ struct __snd_pcm_mmap_status64 { struct __snd_pcm_mmap_control64 { __pad_before_uframe __pad1; snd_pcm_uframes_t appl_ptr; /* RW: appl ptr (0...boundary-1) */ - __pad_before_uframe __pad2; + __pad_before_uframe __pad2; // This should be __pad_after_uframe, but binary + // backwards compatibility constraints prevent a fix. __pad_before_uframe __pad3; snd_pcm_uframes_t avail_min; /* RW: min available frames for wakeup */ diff --git a/include/uapi/sound/emu10k1.h b/include/uapi/sound/emu10k1.h index 1c1f1dd44611..c8e131d6da00 100644 --- a/include/uapi/sound/emu10k1.h +++ b/include/uapi/sound/emu10k1.h @@ -15,9 +15,6 @@ * ---- FX8010 ---- */ -#define EMU10K1_CARD_CREATIVE 0x00000000 -#define EMU10K1_CARD_EMUAPS 0x00000001 - #define EMU10K1_FX8010_PCM_COUNT 8 /* @@ -46,6 +43,19 @@ #define iINTERP 0x0e /* R = A + (X * (Y - A) >> 31) ; saturation */ #define iSKIP 0x0f /* R = A (cc_reg), X (count), Y (cc_test) */ +#define LOWORD_OPX_MASK 0x000ffc00 /* Instruction operand X */ +#define LOWORD_OPY_MASK 0x000003ff /* Instruction operand Y */ +#define HIWORD_OPCODE_MASK 0x00f00000 /* Instruction opcode */ +#define HIWORD_RESULT_MASK 0x000ffc00 /* Instruction result */ +#define HIWORD_OPA_MASK 0x000003ff /* Instruction operand A */ + +/* Audigy Soundcards have a different instruction format */ +#define A_LOWORD_OPX_MASK 0x007ff000 +#define A_LOWORD_OPY_MASK 0x000007ff +#define A_HIWORD_OPCODE_MASK 0x0f000000 +#define A_HIWORD_RESULT_MASK 0x007ff000 +#define A_HIWORD_OPA_MASK 0x000007ff + /* GPRs */ #define FXBUS(x) (0x00 + (x)) /* x = 0x00 - 0x0f */ #define EXTIN(x) (0x10 + (x)) /* x = 0x00 - 0x0f */ @@ -53,6 +63,16 @@ #define FXBUS2(x) (0x30 + (x)) /* x = 0x00 - 0x0f copies of fx buses for capture -> FXWC high 16 bits */ /* NB: 0x31 and 0x32 are shared with Center/LFE on SB live 5.1 */ +#define A_FXBUS(x) (0x00 + (x)) /* x = 0x00 - 0x3f FX buses */ +#define A_EXTIN(x) (0x40 + (x)) /* x = 0x00 - 0x0f physical ins */ +#define A_P16VIN(x) (0x50 + (x)) /* x = 0x00 - 0x0f p16v ins (A2 only) "EMU32 inputs" */ +#define A_EXTOUT(x) (0x60 + (x)) /* x = 0x00 - 0x1f physical outs -> A_FXWC1 0x79-7f unknown */ +#define A_FXBUS2(x) (0x80 + (x)) /* x = 0x00 - 0x1f extra outs used for EFX capture -> A_FXWC2 */ +#define A_EMU32OUTH(x) (0xa0 + (x)) /* x = 0x00 - 0x0f "EMU32_OUT_10 - _1F" */ +#define A_EMU32OUTL(x) (0xb0 + (x)) /* x = 0x00 - 0x0f "EMU32_OUT_01 - _0F" */ +#define A3_EMU32IN(x) (0x160 + (x)) /* x = 0x00 - 0x1f "EMU32_IN_00 - _1F" - Only when .device = 0x0008 */ +#define A3_EMU32OUT(x) (0x1E0 + (x)) /* x = 0x00 - 0x1f "EMU32_OUT_00 - _1F" - Only when .device = 0x0008 */ + #define C_00000000 0x40 #define C_00000001 0x41 #define C_00000002 0x42 @@ -81,12 +101,66 @@ #define GPR_NOISE1 0x59 /* noise source */ #define GPR_IRQ 0x5a /* IRQ register */ #define GPR_DBAC 0x5b /* TRAM Delay Base Address Counter */ + +/* Audigy constants */ +#define A_C_00000000 0xc0 +#define A_C_00000001 0xc1 +#define A_C_00000002 0xc2 +#define A_C_00000003 0xc3 +#define A_C_00000004 0xc4 +#define A_C_00000008 0xc5 +#define A_C_00000010 0xc6 +#define A_C_00000020 0xc7 +#define A_C_00000100 0xc8 +#define A_C_00010000 0xc9 +#define A_C_00000800 0xca +#define A_C_10000000 0xcb +#define A_C_20000000 0xcc +#define A_C_40000000 0xcd +#define A_C_80000000 0xce +#define A_C_7fffffff 0xcf +#define A_C_ffffffff 0xd0 +#define A_C_fffffffe 0xd1 +#define A_C_c0000000 0xd2 +#define A_C_4f1bbcdc 0xd3 +#define A_C_5a7ef9db 0xd4 +#define A_C_00100000 0xd5 +#define A_GPR_ACCU 0xd6 /* ACCUM, accumulator */ +#define A_GPR_COND 0xd7 /* CCR, condition register */ +#define A_GPR_NOISE0 0xd8 /* noise source */ +#define A_GPR_NOISE1 0xd9 /* noise source */ +#define A_GPR_IRQ 0xda /* IRQ register */ +#define A_GPR_DBAC 0xdb /* TRAM Delay Base Address Counter - internal */ +#define A_GPR_DBACE 0xde /* TRAM Delay Base Address Counter - external */ + +/* Each FX general purpose register is 32 bits in length, all bits are used */ +#define FXGPREGBASE 0x100 /* FX general purpose registers base */ +#define A_FXGPREGBASE 0x400 /* Audigy GPRs, 0x400 to 0x5ff */ + +#define A_TANKMEMCTLREGBASE 0x100 /* Tank memory control registers base - only for Audigy */ +#define A_TANKMEMCTLREG_MASK 0x1f /* only 5 bits used - only for Audigy */ + +/* Tank audio data is logarithmically compressed down to 16 bits before writing to TRAM and is */ +/* decompressed back to 20 bits on a read. There are a total of 160 locations, the last 32 */ +/* locations are for external TRAM. */ +#define TANKMEMDATAREGBASE 0x200 /* Tank memory data registers base */ +#define TANKMEMDATAREG_MASK 0x000fffff /* 20 bit tank audio data field */ + +/* Combined address field and memory opcode or flag field. 160 locations, last 32 are external */ +#define TANKMEMADDRREGBASE 0x300 /* Tank memory address registers base */ +#define TANKMEMADDRREG_ADDR_MASK 0x000fffff /* 20 bit tank address field */ +#define TANKMEMADDRREG_CLEAR 0x00800000 /* Clear tank memory */ +#define TANKMEMADDRREG_ALIGN 0x00400000 /* Align read or write relative to tank access */ +#define TANKMEMADDRREG_WRITE 0x00200000 /* Write to tank memory */ +#define TANKMEMADDRREG_READ 0x00100000 /* Read from tank memory */ + #define GPR(x) (FXGPREGBASE + (x)) /* free GPRs: x = 0x00 - 0xff */ #define ITRAM_DATA(x) (TANKMEMDATAREGBASE + 0x00 + (x)) /* x = 0x00 - 0x7f */ #define ETRAM_DATA(x) (TANKMEMDATAREGBASE + 0x80 + (x)) /* x = 0x00 - 0x1f */ #define ITRAM_ADDR(x) (TANKMEMADDRREGBASE + 0x00 + (x)) /* x = 0x00 - 0x7f */ #define ETRAM_ADDR(x) (TANKMEMADDRREGBASE + 0x80 + (x)) /* x = 0x00 - 0x1f */ +#define A_GPR(x) (A_FXGPREGBASE + (x)) #define A_ITRAM_DATA(x) (TANKMEMDATAREGBASE + 0x00 + (x)) /* x = 0x00 - 0xbf */ #define A_ETRAM_DATA(x) (TANKMEMDATAREGBASE + 0xc0 + (x)) /* x = 0x00 - 0x3f */ #define A_ITRAM_ADDR(x) (TANKMEMADDRREGBASE + 0x00 + (x)) /* x = 0x00 - 0xbf */ @@ -94,17 +168,6 @@ #define A_ITRAM_CTL(x) (A_TANKMEMCTLREGBASE + 0x00 + (x)) /* x = 0x00 - 0xbf */ #define A_ETRAM_CTL(x) (A_TANKMEMCTLREGBASE + 0xc0 + (x)) /* x = 0x00 - 0x3f */ -#define A_FXBUS(x) (0x00 + (x)) /* x = 0x00 - 0x3f FX buses */ -#define A_EXTIN(x) (0x40 + (x)) /* x = 0x00 - 0x0f physical ins */ -#define A_P16VIN(x) (0x50 + (x)) /* x = 0x00 - 0x0f p16v ins (A2 only) "EMU32 inputs" */ -#define A_EXTOUT(x) (0x60 + (x)) /* x = 0x00 - 0x1f physical outs -> A_FXWC1 0x79-7f unknown */ -#define A_FXBUS2(x) (0x80 + (x)) /* x = 0x00 - 0x1f extra outs used for EFX capture -> A_FXWC2 */ -#define A_EMU32OUTH(x) (0xa0 + (x)) /* x = 0x00 - 0x0f "EMU32_OUT_10 - _1F" - ??? */ -#define A_EMU32OUTL(x) (0xb0 + (x)) /* x = 0x00 - 0x0f "EMU32_OUT_1 - _F" - ??? */ -#define A3_EMU32IN(x) (0x160 + (x)) /* x = 0x00 - 0x3f "EMU32_IN_00 - _3F" - Only when .device = 0x0008 */ -#define A3_EMU32OUT(x) (0x1E0 + (x)) /* x = 0x00 - 0x0f "EMU32_OUT_00 - _3F" - Only when .device = 0x0008 */ -#define A_GPR(x) (A_FXGPREGBASE + (x)) - /* cc_reg constants */ #define CC_REG_NORMALIZED C_00000001 #define CC_REG_BORROW C_00000002 @@ -113,7 +176,17 @@ #define CC_REG_SATURATE C_00000010 #define CC_REG_NONZERO C_00000100 +#define A_CC_REG_NORMALIZED A_C_00000001 +#define A_CC_REG_BORROW A_C_00000002 +#define A_CC_REG_MINUS A_C_00000004 +#define A_CC_REG_ZERO A_C_00000008 +#define A_CC_REG_SATURATE A_C_00000010 +#define A_CC_REG_NONZERO A_C_00000100 + /* FX buses */ +// These are arbitrary mappings; our DSP code simply expects +// the config files to route the channels this way. +// The numbers are documented in {audigy,sb-live}-mixer.rst. #define FXBUS_PCM_LEFT 0x00 #define FXBUS_PCM_RIGHT 0x01 #define FXBUS_PCM_LEFT_REAR 0x02 @@ -203,38 +276,7 @@ #define A_EXTOUT_ADC_CAP_R 0x17 /* right */ #define A_EXTOUT_MIC_CAP 0x18 /* Mic capture buffer */ -/* Audigy constants */ -#define A_C_00000000 0xc0 -#define A_C_00000001 0xc1 -#define A_C_00000002 0xc2 -#define A_C_00000003 0xc3 -#define A_C_00000004 0xc4 -#define A_C_00000008 0xc5 -#define A_C_00000010 0xc6 -#define A_C_00000020 0xc7 -#define A_C_00000100 0xc8 -#define A_C_00010000 0xc9 -#define A_C_00000800 0xca -#define A_C_10000000 0xcb -#define A_C_20000000 0xcc -#define A_C_40000000 0xcd -#define A_C_80000000 0xce -#define A_C_7fffffff 0xcf -#define A_C_ffffffff 0xd0 -#define A_C_fffffffe 0xd1 -#define A_C_c0000000 0xd2 -#define A_C_4f1bbcdc 0xd3 -#define A_C_5a7ef9db 0xd4 -#define A_C_00100000 0xd5 -#define A_GPR_ACCU 0xd6 /* ACCUM, accumulator */ -#define A_GPR_COND 0xd7 /* CCR, condition register */ -#define A_GPR_NOISE0 0xd8 /* noise source */ -#define A_GPR_NOISE1 0xd9 /* noise source */ -#define A_GPR_IRQ 0xda /* IRQ register */ -#define A_GPR_DBAC 0xdb /* TRAM Delay Base Address Counter - internal */ -#define A_GPR_DBACE 0xde /* TRAM Delay Base Address Counter - external */ - -/* definitions for debug register */ +/* Definitions for debug register. Note that these are for emu10k1 ONLY. */ #define EMU10K1_DBG_ZC 0x80000000 /* zero tram counter */ #define EMU10K1_DBG_SATURATION_OCCURED 0x02000000 /* saturation control */ #define EMU10K1_DBG_SATURATION_ADDR 0x01ff0000 /* saturation address */ @@ -243,14 +285,14 @@ #define EMU10K1_DBG_CONDITION_CODE 0x00003e00 /* condition code */ #define EMU10K1_DBG_SINGLE_STEP_ADDR 0x000001ff /* single step address */ -/* tank memory address line */ -#ifndef __KERNEL__ -#define TANKMEMADDRREG_ADDR_MASK 0x000fffff /* 20 bit tank address field */ -#define TANKMEMADDRREG_CLEAR 0x00800000 /* Clear tank memory */ -#define TANKMEMADDRREG_ALIGN 0x00400000 /* Align read or write relative to tank access */ -#define TANKMEMADDRREG_WRITE 0x00200000 /* Write to tank memory */ -#define TANKMEMADDRREG_READ 0x00100000 /* Read from tank memory */ -#endif +/* Definitions for emu10k2 debug register. */ +#define A_DBG_ZC 0x40000000 /* zero tram counter */ +#define A_DBG_SATURATION_OCCURED 0x20000000 +#define A_DBG_SATURATION_ADDR 0x0ffc0000 +#define A_DBG_SINGLE_STEP 0x00020000 /* Set to zero to start dsp */ +#define A_DBG_STEP 0x00010000 +#define A_DBG_CONDITION_CODE 0x0000f800 +#define A_DBG_STEP_ADDR 0x000003ff struct snd_emu10k1_fx8010_info { unsigned int internal_tram_size; /* in samples */ diff --git a/sound/ac97_bus.c b/sound/ac97_bus.c index b4685c53ff11..c7aee8c42c55 100644 --- a/sound/ac97_bus.c +++ b/sound/ac97_bus.c @@ -75,19 +75,8 @@ int snd_ac97_reset(struct snd_ac97 *ac97, bool try_warm, unsigned int id, } EXPORT_SYMBOL_GPL(snd_ac97_reset); -/* - * Let drivers decide whether they want to support given codec from their - * probe method. Drivers have direct access to the struct snd_ac97 - * structure and may decide based on the id field amongst other things. - */ -static int ac97_bus_match(struct device *dev, struct device_driver *drv) -{ - return 1; -} - struct bus_type ac97_bus_type = { .name = "ac97", - .match = ac97_bus_match, }; static int __init ac97_bus_init(void) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 8b6aeb8a78f7..d21c73944efd 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -42,70 +42,56 @@ static int fill_silence_frames(struct snd_pcm_substream *substream, * * when runtime->silence_size >= runtime->boundary - fill processed area with silence immediately */ -void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_uframes_t new_hw_ptr) +void snd_pcm_playback_silence(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - snd_pcm_uframes_t frames, ofs, transfer; + snd_pcm_uframes_t appl_ptr = READ_ONCE(runtime->control->appl_ptr); + snd_pcm_sframes_t added, hw_avail, frames; + snd_pcm_uframes_t noise_dist, ofs, transfer; int err; + added = appl_ptr - runtime->silence_start; + if (added) { + if (added < 0) + added += runtime->boundary; + if (added < runtime->silence_filled) + runtime->silence_filled -= added; + else + runtime->silence_filled = 0; + runtime->silence_start = appl_ptr; + } + + // This will "legitimately" turn negative on underrun, and will be mangled + // into a huge number by the boundary crossing handling. The initial state + // might also be not quite sane. The code below MUST account for these cases. + hw_avail = appl_ptr - runtime->status->hw_ptr; + if (hw_avail < 0) + hw_avail += runtime->boundary; + + noise_dist = hw_avail + runtime->silence_filled; if (runtime->silence_size < runtime->boundary) { - snd_pcm_sframes_t noise_dist, n; - snd_pcm_uframes_t appl_ptr = READ_ONCE(runtime->control->appl_ptr); - if (runtime->silence_start != appl_ptr) { - n = appl_ptr - runtime->silence_start; - if (n < 0) - n += runtime->boundary; - if ((snd_pcm_uframes_t)n < runtime->silence_filled) - runtime->silence_filled -= n; - else - runtime->silence_filled = 0; - runtime->silence_start = appl_ptr; - } - if (runtime->silence_filled >= runtime->buffer_size) - return; - noise_dist = snd_pcm_playback_hw_avail(runtime) + runtime->silence_filled; - if (noise_dist >= (snd_pcm_sframes_t) runtime->silence_threshold) - return; frames = runtime->silence_threshold - noise_dist; + if (frames <= 0) + return; if (frames > runtime->silence_size) frames = runtime->silence_size; } else { - if (new_hw_ptr == ULONG_MAX) { /* initialization */ - snd_pcm_sframes_t avail = snd_pcm_playback_hw_avail(runtime); - if (avail > runtime->buffer_size) - avail = runtime->buffer_size; - runtime->silence_filled = avail > 0 ? avail : 0; - runtime->silence_start = (runtime->status->hw_ptr + - runtime->silence_filled) % - runtime->boundary; - } else { - ofs = runtime->status->hw_ptr; - frames = new_hw_ptr - ofs; - if ((snd_pcm_sframes_t)frames < 0) - frames += runtime->boundary; - runtime->silence_filled -= frames; - if ((snd_pcm_sframes_t)runtime->silence_filled < 0) { - runtime->silence_filled = 0; - runtime->silence_start = new_hw_ptr; - } else { - runtime->silence_start = ofs; - } - } - frames = runtime->buffer_size - runtime->silence_filled; + frames = runtime->buffer_size - noise_dist; + if (frames <= 0) + return; } + if (snd_BUG_ON(frames > runtime->buffer_size)) return; - if (frames == 0) - return; - ofs = runtime->silence_start % runtime->buffer_size; - while (frames > 0) { + ofs = (runtime->silence_start + runtime->silence_filled) % runtime->buffer_size; + do { transfer = ofs + frames > runtime->buffer_size ? runtime->buffer_size - ofs : frames; err = fill_silence_frames(substream, ofs, transfer); snd_BUG_ON(err < 0); runtime->silence_filled += transfer; frames -= transfer; ofs = 0; - } + } while (frames > 0); snd_pcm_dma_buffer_sync(substream, SNDRV_DMA_SYNC_DEVICE); } @@ -439,10 +425,6 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, return 0; } - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && - runtime->silence_size > 0) - snd_pcm_playback_silence(substream, new_hw_ptr); - if (in_interrupt) { delta = new_hw_ptr - runtime->hw_ptr_interrupt; if (delta < 0) @@ -460,6 +442,10 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, runtime->hw_ptr_wrap += runtime->boundary; } + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + runtime->silence_size > 0) + snd_pcm_playback_silence(substream); + update_audio_tstamp(substream, &curr_tstamp, &audio_tstamp); return snd_pcm_update_state(substream, runtime); @@ -1878,15 +1864,14 @@ static int wait_for_avail(struct snd_pcm_substream *substream, if (substream->wait_time) { wait_time = substream->wait_time; } else { - wait_time = 10; + wait_time = 100; if (runtime->rate) { - long t = runtime->period_size * 2 / - runtime->rate; + long t = runtime->buffer_size * 1100 / runtime->rate; wait_time = max(t, wait_time); } - wait_time = msecs_to_jiffies(wait_time * 1000); } + wait_time = msecs_to_jiffies(wait_time); } for (;;) { @@ -1934,8 +1919,8 @@ static int wait_for_avail(struct snd_pcm_substream *substream, } if (!tout) { pcm_dbg(substream->pcm, - "%s write error (DMA or IRQ trouble?)\n", - is_playback ? "playback" : "capture"); + "%s timeout (DMA or IRQ trouble?)\n", + is_playback ? "playback write" : "capture read"); err = -EIO; break; } @@ -2155,6 +2140,8 @@ int pcm_lib_apply_appl_ptr(struct snd_pcm_substream *substream, ret = substream->ops->ack(substream); if (ret < 0) { runtime->control->appl_ptr = old_appl_ptr; + if (ret == -EPIPE) + __snd_pcm_xrun(substream); return ret; } } diff --git a/sound/core/pcm_local.h b/sound/core/pcm_local.h index ecb21697ae3a..42fe3a4e9154 100644 --- a/sound/core/pcm_local.h +++ b/sound/core/pcm_local.h @@ -29,8 +29,7 @@ int snd_pcm_update_state(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime); int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream); -void snd_pcm_playback_silence(struct snd_pcm_substream *substream, - snd_pcm_uframes_t new_hw_ptr); +void snd_pcm_playback_silence(struct snd_pcm_substream *substream); static inline snd_pcm_uframes_t snd_pcm_avail(struct snd_pcm_substream *substream) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 331380c2438b..3d0c4a5b701b 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -958,7 +958,7 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream, if (snd_pcm_running(substream)) { if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) - snd_pcm_playback_silence(substream, ULONG_MAX); + snd_pcm_playback_silence(substream); err = snd_pcm_update_state(substream, runtime); } snd_pcm_stream_unlock_irq(substream); @@ -1455,7 +1455,7 @@ static void snd_pcm_post_start(struct snd_pcm_substream *substream, __snd_pcm_set_state(runtime, state); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) - snd_pcm_playback_silence(substream, ULONG_MAX); + snd_pcm_playback_silence(substream); snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MSTART); } @@ -1916,7 +1916,7 @@ static void snd_pcm_post_reset(struct snd_pcm_substream *substream, runtime->control->appl_ptr = runtime->status->hw_ptr; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) - snd_pcm_playback_silence(substream, ULONG_MAX); + snd_pcm_playback_silence(substream); snd_pcm_stream_unlock_irq(substream); } @@ -2159,12 +2159,12 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, if (runtime->no_period_wakeup) tout = MAX_SCHEDULE_TIMEOUT; else { - tout = 10; + tout = 100; if (runtime->rate) { - long t = runtime->period_size * 2 / runtime->rate; + long t = runtime->buffer_size * 1100 / runtime->rate; tout = max(t, tout); } - tout = msecs_to_jiffies(tout * 1000); + tout = msecs_to_jiffies(tout); } tout = schedule_timeout(tout); @@ -2187,7 +2187,7 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, result = -ESTRPIPE; else { dev_dbg(substream->pcm->card->dev, - "playback drain error (DMA or IRQ trouble?)\n"); + "playback drain timeout (DMA or IRQ trouble?)\n"); snd_pcm_stop(substream, SNDRV_PCM_STATE_SETUP); result = -EIO; } diff --git a/sound/drivers/portman2x4.c b/sound/drivers/portman2x4.c index a515c13a489f..619e3f594477 100644 --- a/sound/drivers/portman2x4.c +++ b/sound/drivers/portman2x4.c @@ -182,21 +182,11 @@ static inline void portman_write_command(struct portman *pm, u8 value) parport_write_control(pm->pardev->port, value); } -static inline u8 portman_read_command(struct portman *pm) -{ - return parport_read_control(pm->pardev->port); -} - static inline u8 portman_read_status(struct portman *pm) { return parport_read_status(pm->pardev->port); } -static inline u8 portman_read_data(struct portman *pm) -{ - return parport_read_data(pm->pardev->port); -} - static inline void portman_write_data(struct portman *pm, u8 value) { parport_write_data(pm->pardev->port, value); diff --git a/sound/firewire/tascam/tascam-stream.c b/sound/firewire/tascam/tascam-stream.c index 53e094cc411f..dfe783d01d7d 100644 --- a/sound/firewire/tascam/tascam-stream.c +++ b/sound/firewire/tascam/tascam-stream.c @@ -490,7 +490,7 @@ int snd_tscm_stream_start_duplex(struct snd_tscm *tscm, unsigned int rate) // packet is important for media clock recovery. err = amdtp_domain_start(&tscm->domain, tx_init_skip_cycles, true, true); if (err < 0) - return err; + goto error; if (!amdtp_domain_wait_ready(&tscm->domain, READY_TIMEOUT_MS)) { err = -ETIMEDOUT; diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index ae31bb127594..317bdf6dcbef 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -472,6 +472,15 @@ static const struct config_entry config_table[] = { }, #endif +/* Meteor Lake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_METEORLAKE) + /* Meteorlake-P */ + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, + .device = 0x7e28, + }, +#endif + }; static const struct config_entry *snd_intel_dsp_find_config diff --git a/sound/i2c/cs8427.c b/sound/i2c/cs8427.c index 65012af6a36e..f58b14b49045 100644 --- a/sound/i2c/cs8427.c +++ b/sound/i2c/cs8427.c @@ -561,10 +561,13 @@ int snd_cs8427_iec958_active(struct snd_i2c_device *cs8427, int active) if (snd_BUG_ON(!cs8427)) return -ENXIO; chip = cs8427->private_data; - if (active) + if (active) { memcpy(chip->playback.pcm_status, chip->playback.def_status, 24); - chip->playback.pcm_ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; + chip->playback.pcm_ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; + } else { + chip->playback.pcm_ctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE; + } snd_ctl_notify(cs8427->bus->card, SNDRV_CTL_EVENT_MASK_VALUE | SNDRV_CTL_EVENT_MASK_INFO, &chip->playback.pcm_ctl->id); diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index a55836225401..861958451ef5 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -461,7 +461,7 @@ config SND_INDIGODJX will be called snd-indigodjx config SND_EMU10K1 - tristate "Emu10k1 (SB Live!, Audigy, E-mu APS)" + tristate "Emu10k1 (SB Live!, Audigy, E-MU APS/0404/1010/1212/1616/1820)" select FW_LOADER select SND_HWDEP select SND_RAWMIDI @@ -471,7 +471,7 @@ config SND_EMU10K1 depends on ZONE_DMA help Say Y to include support for Sound Blaster PCI 512, Live!, - Audigy and E-mu APS (partially supported) soundcards. + Audigy and E-MU APS/0404/1010/1212/1616/1820 soundcards. The confusing multitude of mixer controls is documented in <file:Documentation/sound/cards/sb-live-mixer.rst> and diff --git a/sound/pci/asihpi/hpi6000.c b/sound/pci/asihpi/hpi6000.c index 88d902997b74..72aa135d69f8 100644 --- a/sound/pci/asihpi/hpi6000.c +++ b/sound/pci/asihpi/hpi6000.c @@ -1253,7 +1253,6 @@ static u16 hpi6000_dsp_block_read32(struct hpi_adapter_obj *pao, int local_count = count; int xfer_size; u32 *pdata = dest; - u32 loop_count = 0; while (local_count) { if (local_count > c6711_burst_size) @@ -1273,7 +1272,6 @@ static u16 hpi6000_dsp_block_read32(struct hpi_adapter_obj *pao, pdata += xfer_size; local_hpi_address += sizeof(u32) * xfer_size; local_count -= xfer_size; - loop_count++; } if (time_out) diff --git a/sound/pci/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c index 27e11b5f70b9..c7d7eff86727 100644 --- a/sound/pci/asihpi/hpi6205.c +++ b/sound/pci/asihpi/hpi6205.c @@ -430,7 +430,7 @@ void HPI_6205(struct hpi_message *phm, struct hpi_response *phr) pao = hpi_find_adapter(phm->adapter_index); } else { /* subsys messages don't address an adapter */ - _HPI_6205(NULL, phm, phr); + phr->error = HPI_ERROR_INVALID_OBJ_INDEX; return; } diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c index 672af4b9597b..b8163f26004a 100644 --- a/sound/pci/emu10k1/emu10k1.c +++ b/sound/pci/emu10k1/emu10k1.c @@ -68,17 +68,6 @@ static const struct pci_device_id snd_emu10k1_ids[] = { { 0, } }; -/* - * Audigy 2 Value notes: - * A_IOCFG Input (GPIO) - * 0x400 = Front analog jack plugged in. (Green socket) - * 0x1000 = Read analog jack plugged in. (Black socket) - * 0x2000 = Center/LFE analog jack plugged in. (Orange socket) - * A_IOCFG Output (GPIO) - * 0x60 = Sound out of front Left. - * Win sets it to 0xXX61 - */ - MODULE_DEVICE_TABLE(pci, snd_emu10k1_ids); static int snd_card_emu10k1_probe(struct pci_dev *pci, diff --git a/sound/pci/emu10k1/emu10k1_callback.c b/sound/pci/emu10k1/emu10k1_callback.c index dba1e9fc2eec..9455df18f7b2 100644 --- a/sound/pci/emu10k1/emu10k1_callback.c +++ b/sound/pci/emu10k1/emu10k1_callback.c @@ -120,9 +120,9 @@ release_voice(struct snd_emux_voice *vp) struct snd_emu10k1 *hw; hw = vp->hw; - dcysusv = 0x8000 | (unsigned char)vp->reg.parm.modrelease; + dcysusv = (unsigned char)vp->reg.parm.modrelease | DCYSUSM_PHASE1_MASK; snd_emu10k1_ptr_write(hw, DCYSUSM, vp->ch, dcysusv); - dcysusv = 0x8000 | (unsigned char)vp->reg.parm.volrelease | DCYSUSV_CHANNELENABLE_MASK; + dcysusv = (unsigned char)vp->reg.parm.volrelease | DCYSUSV_PHASE1_MASK | DCYSUSV_CHANNELENABLE_MASK; snd_emu10k1_ptr_write(hw, DCYSUSV, vp->ch, dcysusv); } @@ -138,7 +138,8 @@ terminate_voice(struct snd_emux_voice *vp) if (snd_BUG_ON(!vp)) return; hw = vp->hw; - snd_emu10k1_ptr_write(hw, DCYSUSV, vp->ch, 0x807f | DCYSUSV_CHANNELENABLE_MASK); + snd_emu10k1_ptr_write(hw, DCYSUSV, vp->ch, + DCYSUSV_PHASE1_MASK | DCYSUSV_DECAYTIME_MASK | DCYSUSV_CHANNELENABLE_MASK); if (vp->block) { struct snd_emu10k1_memblk *emem; emem = (struct snd_emu10k1_memblk *)vp->block; @@ -347,9 +348,9 @@ start_voice(struct snd_emux_voice *vp) } /* channel to be silent and idle */ - snd_emu10k1_ptr_write(hw, DCYSUSV, ch, 0x0000); - snd_emu10k1_ptr_write(hw, VTFT, ch, 0x0000FFFF); - snd_emu10k1_ptr_write(hw, CVCF, ch, 0x0000FFFF); + snd_emu10k1_ptr_write(hw, DCYSUSV, ch, 0); + snd_emu10k1_ptr_write(hw, VTFT, ch, VTFT_FILTERTARGET_MASK); + snd_emu10k1_ptr_write(hw, CVCF, ch, CVCF_CURRENTFILTER_MASK); snd_emu10k1_ptr_write(hw, PTRX, ch, 0); snd_emu10k1_ptr_write(hw, CPF, ch, 0); @@ -453,7 +454,7 @@ start_voice(struct snd_emux_voice *vp) /* reset volume */ temp = (unsigned int)vp->vtarget << 16; snd_emu10k1_ptr_write(hw, VTFT, ch, temp | vp->ftarget); - snd_emu10k1_ptr_write(hw, CVCF, ch, temp | 0xff00); + snd_emu10k1_ptr_write(hw, CVCF, ch, temp | CVCF_CURRENTFILTER_MASK); return 0; } @@ -531,8 +532,5 @@ set_fm2frq2(struct snd_emu10k1 *hw, struct snd_emux_voice *vp) static void set_filterQ(struct snd_emu10k1 *hw, struct snd_emux_voice *vp) { - unsigned int val; - val = snd_emu10k1_ptr_read(hw, CCCA, vp->ch) & ~CCCA_RESONANCE; - val |= (vp->reg.parm.filterQ << 28); - snd_emu10k1_ptr_write(hw, CCCA, vp->ch, val); + snd_emu10k1_ptr_write(hw, CCCA_RESONANCE, vp->ch, vp->reg.parm.filterQ); } diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 3880f359e688..192208c291d6 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -59,8 +59,8 @@ void snd_emu10k1_voice_init(struct snd_emu10k1 *emu, int ch) { snd_emu10k1_ptr_write(emu, DCYSUSV, ch, 0); snd_emu10k1_ptr_write(emu, IP, ch, 0); - snd_emu10k1_ptr_write(emu, VTFT, ch, 0xffff); - snd_emu10k1_ptr_write(emu, CVCF, ch, 0xffff); + snd_emu10k1_ptr_write(emu, VTFT, ch, VTFT_FILTERTARGET_MASK); + snd_emu10k1_ptr_write(emu, CVCF, ch, CVCF_CURRENTFILTER_MASK); snd_emu10k1_ptr_write(emu, PTRX, ch, 0); snd_emu10k1_ptr_write(emu, CPF, ch, 0); snd_emu10k1_ptr_write(emu, CCR, ch, 0); @@ -74,7 +74,7 @@ void snd_emu10k1_voice_init(struct snd_emu10k1 *emu, int ch) snd_emu10k1_ptr_write(emu, ATKHLDM, ch, 0); snd_emu10k1_ptr_write(emu, DCYSUSM, ch, 0); - snd_emu10k1_ptr_write(emu, IFATN, ch, 0xffff); + snd_emu10k1_ptr_write(emu, IFATN, ch, IFATN_FILTERCUTOFF_MASK | IFATN_ATTENUATION_MASK); snd_emu10k1_ptr_write(emu, PEFE, ch, 0); snd_emu10k1_ptr_write(emu, FMMOD, ch, 0); snd_emu10k1_ptr_write(emu, TREMFRQ, ch, 24); /* 1 Hz */ @@ -90,10 +90,10 @@ void snd_emu10k1_voice_init(struct snd_emu10k1 *emu, int ch) /* Audigy extra stuffs */ if (emu->audigy) { - snd_emu10k1_ptr_write(emu, 0x4c, ch, 0); /* ?? */ - snd_emu10k1_ptr_write(emu, 0x4d, ch, 0); /* ?? */ - snd_emu10k1_ptr_write(emu, 0x4e, ch, 0); /* ?? */ - snd_emu10k1_ptr_write(emu, 0x4f, ch, 0); /* ?? */ + snd_emu10k1_ptr_write(emu, A_CSBA, ch, 0); + snd_emu10k1_ptr_write(emu, A_CSDC, ch, 0); + snd_emu10k1_ptr_write(emu, A_CSFE, ch, 0); + snd_emu10k1_ptr_write(emu, A_CSHG, ch, 0); snd_emu10k1_ptr_write(emu, A_FXRT1, ch, 0x03020100); snd_emu10k1_ptr_write(emu, A_FXRT2, ch, 0x3f3f3f3f); snd_emu10k1_ptr_write(emu, A_SENDAMOUNTS, ch, 0); @@ -140,7 +140,7 @@ static const unsigned int i2c_adc_init[][2] = { { 0x15, ADC_MUX_2 }, /* ADC Mixer control. Mic for A2ZS Notebook */ }; -static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume) +static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir) { unsigned int silent_page; int ch; @@ -162,6 +162,8 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume) outl(0, emu->port + INTE); snd_emu10k1_ptr_write(emu, CLIEL, 0, 0); snd_emu10k1_ptr_write(emu, CLIEH, 0, 0); + + /* disable stop on loop end */ snd_emu10k1_ptr_write(emu, SOLEL, 0, 0); snd_emu10k1_ptr_write(emu, SOLEH, 0, 0); @@ -181,13 +183,11 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume) snd_emu10k1_ptr_write(emu, SPCS1, 0, emu->spdif_bits[1]); snd_emu10k1_ptr_write(emu, SPCS2, 0, emu->spdif_bits[2]); - if (emu->card_capabilities->ca0151_chip) { /* audigy2 */ + if (emu->card_capabilities->emu_model) { + } else if (emu->card_capabilities->ca0151_chip) { /* audigy2 */ /* Hacks for Alice3 to work independent of haP16V driver */ /* Setup SRCMulti_I2S SamplingRate */ - tmp = snd_emu10k1_ptr_read(emu, A_SPDIF_SAMPLERATE, 0); - tmp &= 0xfffff1ff; - tmp |= (0x2<<9); - snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, 0, tmp); + snd_emu10k1_ptr_write(emu, A_I2S_CAPTURE_RATE, 0, A_I2S_CAPTURE_96000); /* Setup SRCSel (Enable Spdif,I2S SRCMulti) */ snd_emu10k1_ptr20_write(emu, SRCSel, 0, 0x14); @@ -199,32 +199,26 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume) outl(0x0201, emu->port + HCFG2); /* Set playback routing. */ snd_emu10k1_ptr20_write(emu, CAPTURE_P16V_SOURCE, 0, 0x78e4); - } - if (emu->card_capabilities->ca0108_chip) { /* audigy2 Value */ + } else if (emu->card_capabilities->ca0108_chip) { /* audigy2 Value */ /* Hacks for Alice3 to work independent of haP16V driver */ dev_info(emu->card->dev, "Audigy2 value: Special config.\n"); /* Setup SRCMulti_I2S SamplingRate */ - tmp = snd_emu10k1_ptr_read(emu, A_SPDIF_SAMPLERATE, 0); - tmp &= 0xfffff1ff; - tmp |= (0x2<<9); - snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, 0, tmp); + snd_emu10k1_ptr_write(emu, A_I2S_CAPTURE_RATE, 0, A_I2S_CAPTURE_96000); /* Setup SRCSel (Enable Spdif,I2S SRCMulti) */ - outl(0x600000, emu->port + 0x20); - outl(0x14, emu->port + 0x24); + snd_emu10k1_ptr20_write(emu, P17V_SRCSel, 0, 0x14); /* Setup SRCMulti Input Audio Enable */ - outl(0x7b0000, emu->port + 0x20); - outl(0xFF000000, emu->port + 0x24); + snd_emu10k1_ptr20_write(emu, P17V_MIXER_I2S_ENABLE, 0, 0xFF000000); /* Setup SPDIF Out Audio Enable */ /* The Audigy 2 Value has a separate SPDIF out, * so no need for a mixer switch */ - outl(0x7a0000, emu->port + 0x20); - outl(0xFF000000, emu->port + 0x24); - tmp = inl(emu->port + A_IOCFG) & ~0x8; /* Clear bit 3 */ - outl(tmp, emu->port + A_IOCFG); + snd_emu10k1_ptr20_write(emu, P17V_MIXER_SPDIF_ENABLE, 0, 0xFF000000); + + tmp = inw(emu->port + A_IOCFG) & ~0x8; /* Clear bit 3 */ + outw(tmp, emu->port + A_IOCFG); } if (emu->card_capabilities->spi_dac) { /* Audigy 2 ZS Notebook with DAC Wolfson WM8768/WM8568 */ int size, n; @@ -244,15 +238,15 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume) * GPIO6: Unknown * GPIO7: Unknown */ - outl(0x76, emu->port + A_IOCFG); /* Windows uses 0x3f76 */ + outw(0x76, emu->port + A_IOCFG); /* Windows uses 0x3f76 */ } if (emu->card_capabilities->i2c_adc) { /* Audigy 2 ZS Notebook with ADC Wolfson WM8775 */ int size, n; snd_emu10k1_ptr20_write(emu, P17V_I2S_SRC_SEL, 0, 0x2020205f); - tmp = inl(emu->port + A_IOCFG); - outl(tmp | 0x4, emu->port + A_IOCFG); /* Set bit 2 for mic input */ - tmp = inl(emu->port + A_IOCFG); + tmp = inw(emu->port + A_IOCFG); + outw(tmp | 0x4, emu->port + A_IOCFG); /* Set bit 2 for mic input */ + tmp = inw(emu->port + A_IOCFG); size = ARRAY_SIZE(i2c_adc_init); for (n = 0; n < size; n++) snd_emu10k1_i2c_write(emu, i2c_adc_init[n][0], i2c_adc_init[n][1]); @@ -265,7 +259,7 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume) snd_emu10k1_ptr_write(emu, PTB, 0, emu->ptb_pages.addr); snd_emu10k1_ptr_write(emu, TCB, 0, 0); /* taken from original driver */ - snd_emu10k1_ptr_write(emu, TCBS, 0, 4); /* taken from original driver */ + snd_emu10k1_ptr_write(emu, TCBS, 0, TCBS_BUFFSIZE_256K); /* taken from original driver */ silent_page = (emu->silent_page.addr << emu->address_mode) | (emu->address_mode ? MAP_PTI_MASK1 : MAP_PTI_MASK0); for (ch = 0; ch < NUM_G; ch++) { @@ -308,12 +302,12 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume) } else if (emu->card_capabilities->i2c_adc) { ; /* Disable A_IOCFG for Audigy 2 ZS Notebook */ } else if (emu->audigy) { - unsigned int reg = inl(emu->port + A_IOCFG); - outl(reg | A_IOCFG_GPOUT2, emu->port + A_IOCFG); + u16 reg = inw(emu->port + A_IOCFG); + outw(reg | A_IOCFG_GPOUT2, emu->port + A_IOCFG); udelay(500); - outl(reg | A_IOCFG_GPOUT1 | A_IOCFG_GPOUT2, emu->port + A_IOCFG); + outw(reg | A_IOCFG_GPOUT1 | A_IOCFG_GPOUT2, emu->port + A_IOCFG); udelay(100); - outl(reg, emu->port + A_IOCFG); + outw(reg, emu->port + A_IOCFG); } else { unsigned int reg = inl(emu->port + HCFG); outl(reg | HCFG_GPOUT2, emu->port + HCFG); @@ -329,8 +323,8 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume) } else if (emu->card_capabilities->i2c_adc) { ; /* Disable A_IOCFG for Audigy 2 ZS Notebook */ } else if (emu->audigy) { /* enable analog output */ - unsigned int reg = inl(emu->port + A_IOCFG); - outl(reg | A_IOCFG_GPOUT0, emu->port + A_IOCFG); + u16 reg = inw(emu->port + A_IOCFG); + outw(reg | A_IOCFG_GPOUT0, emu->port + A_IOCFG); } if (emu->address_mode == 0) { @@ -354,19 +348,19 @@ static void snd_emu10k1_audio_enable(struct snd_emu10k1 *emu) } else if (emu->card_capabilities->i2c_adc) { ; /* Disable A_IOCFG for Audigy 2 ZS Notebook */ } else if (emu->audigy) { - outl(inl(emu->port + A_IOCFG) & ~0x44, emu->port + A_IOCFG); + outw(inw(emu->port + A_IOCFG) & ~0x44, emu->port + A_IOCFG); if (emu->card_capabilities->ca0151_chip) { /* audigy2 */ /* Unmute Analog now. Set GPO6 to 1 for Apollo. * This has to be done after init ALice3 I2SOut beyond 48KHz. * So, sequence is important. */ - outl(inl(emu->port + A_IOCFG) | 0x0040, emu->port + A_IOCFG); + outw(inw(emu->port + A_IOCFG) | 0x0040, emu->port + A_IOCFG); } else if (emu->card_capabilities->ca0108_chip) { /* audigy2 value */ /* Unmute Analog now. */ - outl(inl(emu->port + A_IOCFG) | 0x0060, emu->port + A_IOCFG); + outw(inw(emu->port + A_IOCFG) | 0x0060, emu->port + A_IOCFG); } else { /* Disable routing from AC97 line out to Front speakers */ - outl(inl(emu->port + A_IOCFG) | 0x0080, emu->port + A_IOCFG); + outw(inw(emu->port + A_IOCFG) | 0x0080, emu->port + A_IOCFG); } } @@ -651,26 +645,27 @@ static int snd_emu1010_load_firmware_entry(struct snd_emu10k1 *emu, const struct firmware *fw_entry) { int n, i; - int reg; - int value; - __always_unused unsigned int write_post; + u16 reg; + u8 value; + __always_unused u16 write_post; unsigned long flags; if (!fw_entry) return -EIO; /* The FPGA is a Xilinx Spartan IIE XC2S50E */ + /* On E-MU 0404b it is a Xilinx Spartan III XC3S50 */ /* GPIO7 -> FPGA PGMN * GPIO6 -> FPGA CCLK * GPIO5 -> FPGA DIN * FPGA CONFIG OFF -> FPGA PGMN */ spin_lock_irqsave(&emu->emu_lock, flags); - outl(0x00, emu->port + A_IOCFG); /* Set PGMN low for 1uS. */ - write_post = inl(emu->port + A_IOCFG); + outw(0x00, emu->port + A_GPIO); /* Set PGMN low for 100uS. */ + write_post = inw(emu->port + A_GPIO); udelay(100); - outl(0x80, emu->port + A_IOCFG); /* Leave bit 7 set during netlist setup. */ - write_post = inl(emu->port + A_IOCFG); + outw(0x80, emu->port + A_GPIO); /* Leave bit 7 set during netlist setup. */ + write_post = inw(emu->port + A_GPIO); udelay(100); /* Allow FPGA memory to clean */ for (n = 0; n < fw_entry->size; n++) { value = fw_entry->data[n]; @@ -679,15 +674,15 @@ static int snd_emu1010_load_firmware_entry(struct snd_emu10k1 *emu, if (value & 0x1) reg = reg | 0x20; value = value >> 1; - outl(reg, emu->port + A_IOCFG); - write_post = inl(emu->port + A_IOCFG); - outl(reg | 0x40, emu->port + A_IOCFG); - write_post = inl(emu->port + A_IOCFG); + outw(reg, emu->port + A_GPIO); + write_post = inw(emu->port + A_GPIO); + outw(reg | 0x40, emu->port + A_GPIO); + write_post = inw(emu->port + A_GPIO); } } /* After programming, set GPIO bit 4 high again. */ - outl(0x10, emu->port + A_IOCFG); - write_post = inl(emu->port + A_IOCFG); + outw(0x10, emu->port + A_GPIO); + write_post = inw(emu->port + A_GPIO); spin_unlock_irqrestore(&emu->emu_lock, flags); return 0; @@ -782,7 +777,7 @@ static void emu1010_firmware_work(struct work_struct *work) } else if (!reg && emu->emu1010.last_reg) { /* Audio Dock removed */ dev_info(emu->card->dev, "emu1010: Audio Dock detached\n"); - /* Unmute all */ + /* The hardware auto-mutes all, so we unmute again */ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE); } @@ -794,29 +789,6 @@ static void emu1010_firmware_work(struct work_struct *work) } /* - * EMU-1010 - details found out from this driver, official MS Win drivers, - * testing the card: - * - * Audigy2 (aka Alice2): - * --------------------- - * * communication over PCI - * * conversion of 32-bit data coming over EMU32 links from HANA FPGA - * to 2 x 16-bit, using internal DSP instructions - * * slave mode, clock supplied by HANA - * * linked to HANA using: - * 32 x 32-bit serial EMU32 output channels - * 16 x EMU32 input channels - * (?) x I2S I/O channels (?) - * - * FPGA (aka HANA): - * --------------- - * * provides all (?) physical inputs and outputs of the card - * (ADC, DAC, SPDIF I/O, ADAT I/O, etc.) - * * provides clock signal for the card and Alice2 - * * two crystals - for 44.1kHz and 48kHz multiples - * * provides internal routing of signal sources to signal destinations - * * inputs/outputs to Alice2 - see above - * * Current status of the driver: * ---------------------------- * * only 44.1/48kHz supported (the MS Win driver supports up to 192 kHz) @@ -831,24 +803,10 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 *emu) int err; dev_info(emu->card->dev, "emu1010: Special config.\n"); - /* AC97 2.1, Any 16Meg of 4Gig address, Auto-Mute, EMU32 Slave, - * Lock Sound Memory Cache, Lock Tank Memory Cache, - * Mute all codecs. - */ - outl(0x0005a00c, emu->port + HCFG); - /* AC97 2.1, Any 16Meg of 4Gig address, Auto-Mute, EMU32 Slave, - * Lock Tank Memory Cache, - * Mute all codecs. - */ - outl(0x0005a004, emu->port + HCFG); - /* AC97 2.1, Any 16Meg of 4Gig address, Auto-Mute, EMU32 Slave, - * Mute all codecs. - */ - outl(0x0005a000, emu->port + HCFG); - /* AC97 2.1, Any 16Meg of 4Gig address, Auto-Mute, EMU32 Slave, - * Mute all codecs. - */ - outl(0x0005a000, emu->port + HCFG); + + /* Mute, and disable audio and lock cache, just in case. + * Proper init follows in snd_emu10k1_init(). */ + outl(HCFG_LOCKSOUNDCACHE | HCFG_LOCKTANKCACHE_MASK, emu->port + HCFG); /* Disable 48Volt power to Audio Dock */ snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, 0); @@ -860,7 +818,7 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 *emu) /* FPGA netlist already present so clear it */ /* Return to programming mode */ - snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, 0x02); + snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, EMU_HANA_FPGA_CONFIG_HANA); } snd_emu1010_fpga_read(emu, EMU_HANA_ID, ®); dev_dbg(emu->card->dev, "reg2 = 0x%x\n", reg); @@ -897,54 +855,39 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 *emu) snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, ®); dev_info(emu->card->dev, "emu1010: Card options = 0x%x\n", reg); - snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, ®); - dev_info(emu->card->dev, "emu1010: Card options = 0x%x\n", reg); - snd_emu1010_fpga_read(emu, EMU_HANA_OPTICAL_TYPE, &tmp); /* Optical -> ADAT I/O */ - /* 0 : SPDIF - * 1 : ADAT - */ emu->emu1010.optical_in = 1; /* IN_ADAT */ - emu->emu1010.optical_out = 1; /* IN_ADAT */ - tmp = 0; - tmp = (emu->emu1010.optical_in ? EMU_HANA_OPTICAL_IN_ADAT : 0) | - (emu->emu1010.optical_out ? EMU_HANA_OPTICAL_OUT_ADAT : 0); + emu->emu1010.optical_out = 1; /* OUT_ADAT */ + tmp = (emu->emu1010.optical_in ? EMU_HANA_OPTICAL_IN_ADAT : EMU_HANA_OPTICAL_IN_SPDIF) | + (emu->emu1010.optical_out ? EMU_HANA_OPTICAL_OUT_ADAT : EMU_HANA_OPTICAL_OUT_SPDIF); snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, tmp); - snd_emu1010_fpga_read(emu, EMU_HANA_ADC_PADS, &tmp); /* Set no attenuation on Audio Dock pads. */ - snd_emu1010_fpga_write(emu, EMU_HANA_ADC_PADS, 0x00); emu->emu1010.adc_pads = 0x00; - snd_emu1010_fpga_read(emu, EMU_HANA_DOCK_MISC, &tmp); + snd_emu1010_fpga_write(emu, EMU_HANA_ADC_PADS, emu->emu1010.adc_pads); /* Unmute Audio dock DACs, Headphone source DAC-4. */ - snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_MISC, 0x30); - snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, 0x12); - snd_emu1010_fpga_read(emu, EMU_HANA_DAC_PADS, &tmp); + snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_MISC, EMU_HANA_DOCK_PHONES_192_DAC4); /* DAC PADs. */ - snd_emu1010_fpga_write(emu, EMU_HANA_DAC_PADS, 0x0f); - emu->emu1010.dac_pads = 0x0f; - snd_emu1010_fpga_read(emu, EMU_HANA_DOCK_MISC, &tmp); - snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_MISC, 0x30); - snd_emu1010_fpga_read(emu, EMU_HANA_SPDIF_MODE, &tmp); + emu->emu1010.dac_pads = EMU_HANA_DOCK_DAC_PAD1 | EMU_HANA_DOCK_DAC_PAD2 | + EMU_HANA_DOCK_DAC_PAD3 | EMU_HANA_DOCK_DAC_PAD4; + snd_emu1010_fpga_write(emu, EMU_HANA_DAC_PADS, emu->emu1010.dac_pads); /* SPDIF Format. Set Consumer mode, 24bit, copy enable */ - snd_emu1010_fpga_write(emu, EMU_HANA_SPDIF_MODE, 0x10); + snd_emu1010_fpga_write(emu, EMU_HANA_SPDIF_MODE, EMU_HANA_SPDIF_MODE_RX_INVALID); /* MIDI routing */ - snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_IN, 0x19); - /* Unknown. */ - snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_OUT, 0x0c); + snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_IN, EMU_HANA_MIDI_INA_FROM_HAMOA | EMU_HANA_MIDI_INB_FROM_DOCK2); + snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_OUT, EMU_HANA_MIDI_OUT_DOCK2 | EMU_HANA_MIDI_OUT_SYNC2); /* IRQ Enable: All on */ - /* snd_emu1010_fpga_write(emu, 0x09, 0x0f ); */ + /* snd_emu1010_fpga_write(emu, EMU_HANA_IRQ_ENABLE, 0x0f); */ /* IRQ Enable: All off */ snd_emu1010_fpga_write(emu, EMU_HANA_IRQ_ENABLE, 0x00); - snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, ®); - dev_info(emu->card->dev, "emu1010: Card options3 = 0x%x\n", reg); + emu->emu1010.internal_clock = 1; /* 48000 */ /* Default WCLK set to 48kHz. */ - snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, 0x00); + snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, EMU_HANA_DEFCLOCK_48K); /* Word Clock source, Internal 48kHz x1 */ snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K); /* snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K | EMU_HANA_WCLOCK_4X); */ /* Audio Dock LEDs. */ - snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, 0x12); + snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, EMU_HANA_DOCK_LEDS_2_LOCK | EMU_HANA_DOCK_LEDS_2_48K); #if 0 /* For 96kHz */ @@ -1071,30 +1014,7 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 *emu) EMU_DST_ALICE_I2S2_LEFT, EMU_SRC_DOCK_ADC3_LEFT1); snd_emu1010_fpga_link_dst_src_write(emu, EMU_DST_ALICE_I2S2_RIGHT, EMU_SRC_DOCK_ADC3_RIGHT1); - snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, 0x01); /* Unmute all */ - - snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &tmp); - - /* AC97 1.03, Any 32Meg of 2Gig address, Auto-Mute, EMU32 Slave, - * Lock Sound Memory Cache, Lock Tank Memory Cache, - * Mute all codecs. - */ - outl(0x0000a000, emu->port + HCFG); - /* AC97 1.03, Any 32Meg of 2Gig address, Auto-Mute, EMU32 Slave, - * Lock Sound Memory Cache, Lock Tank Memory Cache, - * Un-Mute all codecs. - */ - outl(0x0000a001, emu->port + HCFG); - - /* Initial boot complete. Now patches */ - - snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &tmp); - snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_IN, 0x19); /* MIDI Route */ - snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_OUT, 0x0c); /* Unknown */ - snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_IN, 0x19); /* MIDI Route */ - snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_OUT, 0x0c); /* Unknown */ - snd_emu1010_fpga_read(emu, EMU_HANA_SPDIF_MODE, &tmp); - snd_emu1010_fpga_write(emu, EMU_HANA_SPDIF_MODE, 0x10); /* SPDIF Format spdif (or 0x11 for aes/ebu) */ + snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE); #if 0 snd_emu1010_fpga_link_dst_src_write(emu, @@ -1215,21 +1135,6 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 *emu) EMU_DST_HANA_ADAT + 7, EMU_SRC_ALICE_EMU32A + 7); emu->emu1010.output_source[23] = 28; } - /* TEMP: Select SPDIF in/out */ - /* snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, 0x0); */ /* Output spdif */ - - /* TEMP: Select 48kHz SPDIF out */ - snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, 0x0); /* Mute all */ - snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, 0x0); /* Default fallback clock 48kHz */ - /* Word Clock source, Internal 48kHz x1 */ - snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K); - /* snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K | EMU_HANA_WCLOCK_4X); */ - emu->emu1010.internal_clock = 1; /* 48000 */ - snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, 0x12); /* Set LEDs on Audio Dock */ - snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, 0x1); /* Unmute all */ - /* snd_emu1010_fpga_write(emu, 0x7, 0x0); */ /* Mute all */ - /* snd_emu1010_fpga_write(emu, 0x7, 0x1); */ /* Unmute all */ - /* snd_emu1010_fpga_write(emu, 0xe, 0x12); */ /* Set LEDs on Audio Dock */ return 0; } @@ -1343,6 +1248,15 @@ static const struct snd_emu_chip_details emu_chip_details[] = { * AC97: STAC9750 * CA0151: None */ + /* + * A_IOCFG Input (GPIO) + * 0x400 = Front analog jack plugged in. (Green socket) + * 0x1000 = Rear analog jack plugged in. (Black socket) + * 0x2000 = Center/LFE analog jack plugged in. (Orange socket) + * A_IOCFG Output (GPIO) + * 0x60 = Sound out of front Left. + * Win sets it to 0xXX61 + */ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x10011102, .driver = "Audigy2", .name = "SB Audigy 2 Value [SB0400]", .id = "Audigy2", @@ -1391,6 +1305,9 @@ static const struct snd_emu_chip_details emu_chip_details[] = { .spi_dac = 1, .i2c_adc = 1, .spk71 = 1} , + /* This is MAEM8950 "Mana" */ + /* Attach MicroDock[M] to make it an E-MU 1616[m]. */ + /* Does NOT support sync daughter card (obviously). */ /* Tested by James@superbug.co.uk 4th Nov 2007. */ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x42011102, .driver = "Audigy2", .name = "E-mu 1010 Notebook [MAEM8950]", @@ -1401,7 +1318,10 @@ static const struct snd_emu_chip_details emu_chip_details[] = { .spk71 = 1 , .emu_model = EMU_MODEL_EMU1616}, /* Tested by James@superbug.co.uk 4th Nov 2007. */ - /* This is MAEM8960, 0202 is MAEM 8980 */ + /* This is MAEM8960 "Hana3", 0202 is MAEM8980 */ + /* Attach 0202 daughter card to make it an E-MU 1212m, OR a + * MicroDock[M] to make it an E-MU 1616[m]. */ + /* Does NOT support sync daughter card. */ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40041102, .driver = "Audigy2", .name = "E-mu 1010b PCI [MAEM8960]", .id = "EMU1010", @@ -1411,6 +1331,11 @@ static const struct snd_emu_chip_details emu_chip_details[] = { .emu_model = EMU_MODEL_EMU1010B}, /* EMU 1010 new revision */ /* Tested by Maxim Kachur <mcdebugger@duganet.ru> 17th Oct 2012. */ /* This is MAEM8986, 0202 is MAEM8980 */ + /* Attach 0202 daughter card to make it an E-MU 1212m, OR a + * MicroDockM to make it an E-MU 1616m. The non-m + * version was never sold with this card, but should + * still work. */ + /* Does NOT support sync daughter card. */ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40071102, .driver = "Audigy2", .name = "E-mu 1010 PCIe [MAEM8986]", .id = "EMU1010", @@ -1419,7 +1344,10 @@ static const struct snd_emu_chip_details emu_chip_details[] = { .spk71 = 1, .emu_model = EMU_MODEL_EMU1010B}, /* EMU 1010 PCIe */ /* Tested by James@superbug.co.uk 8th July 2005. */ - /* This is MAEM8810, 0202 is MAEM8820 */ + /* This is MAEM8810 "Hana", 0202 is MAEM8820 "Hamoa" */ + /* Attach 0202 daughter card to make it an E-MU 1212m, OR an + * AudioDock[M] to make it an E-MU 1820[m]. */ + /* Supports sync daughter card. */ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x40011102, .driver = "Audigy2", .name = "E-mu 1010 [MAEM8810]", .id = "EMU1010", @@ -1427,7 +1355,9 @@ static const struct snd_emu_chip_details emu_chip_details[] = { .ca0102_chip = 1, .spk71 = 1, .emu_model = EMU_MODEL_EMU1010}, /* EMU 1010 old revision */ - /* EMU0404b */ + /* This is MAEM8852 "HanaLiteLite" */ + /* Supports sync daughter card. */ + /* Tested by oswald.buddenhagen@gmx.de Mar 2023. */ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40021102, .driver = "Audigy2", .name = "E-mu 0404b PCI [MAEM8852]", .id = "EMU0404", @@ -1435,6 +1365,8 @@ static const struct snd_emu_chip_details emu_chip_details[] = { .ca0108_chip = 1, .spk71 = 1, .emu_model = EMU_MODEL_EMU0404}, /* EMU 0404 new revision */ + /* This is MAEM8850 "HanaLite" */ + /* Supports sync daughter card. */ /* Tested by James@superbug.co.uk 20-3-2007. */ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x40021102, .driver = "Audigy2", .name = "E-mu 0404 [MAEM8850]", @@ -1444,6 +1376,7 @@ static const struct snd_emu_chip_details emu_chip_details[] = { .spk71 = 1, .emu_model = EMU_MODEL_EMU0404}, /* EMU 0404 */ /* EMU0404 PCIe */ + /* Does NOT support sync daughter card. */ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40051102, .driver = "Audigy2", .name = "E-mu 0404 PCIe [MAEM8984]", .id = "EMU0404", @@ -1451,7 +1384,6 @@ static const struct snd_emu_chip_details emu_chip_details[] = { .ca0108_chip = 1, .spk71 = 1, .emu_model = EMU_MODEL_EMU0404}, /* EMU 0404 PCIe ver_03 */ - /* Note that all E-mu cards require kernel 2.6 or newer. */ {.vendor = 0x1102, .device = 0x0008, .driver = "Audigy2", .name = "SB Audigy 2 Value [Unknown]", .id = "Audigy2", @@ -1532,6 +1464,8 @@ static const struct snd_emu_chip_details emu_chip_details[] = { .spdif_bug = 1, .adc_1361t = 1, /* 24 bit capture instead of 16bit */ .ac97_chip = 1} , + /* Audigy 2 Platinum EX */ + /* Win driver sets A_IOCFG output to 0x1c00 */ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10051102, .driver = "Audigy2", .name = "Audigy 2 Platinum EX [SB0280]", .id = "Audigy2", @@ -1552,6 +1486,8 @@ static const struct snd_emu_chip_details emu_chip_details[] = { .spdif_bug = 1, .invert_shared_spdif = 1, /* digital/analog switch swapped */ .ac97_chip = 1} , + /* Audigy 2 Platinum */ + /* Win driver sets A_IOCFG output to 0xa00 */ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10021102, .driver = "Audigy2", .name = "SB Audigy 2 Platinum [SB0240P]", .id = "Audigy2", @@ -1657,6 +1593,9 @@ static const struct snd_emu_chip_details emu_chip_details[] = { .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , + /* SB Live! Platinum */ + /* Win driver sets A_IOCFG output to 0 */ + /* Tested by Jonathan Dowland <jon@dow.land> Apr 2023. */ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80401102, .driver = "EMU10K1", .name = "SB Live! Platinum [CT4760P]", .id = "Live", @@ -1901,11 +1840,12 @@ int snd_emu10k1_create(struct snd_card *card, pci_set_master(pci); - emu->fx8010.fxbus_mask = 0x303f; + // The masks are not used for Audigy. + // FIXME: these should come from the card_capabilites table. if (extin_mask == 0) - extin_mask = 0x3fcf; + extin_mask = 0x3fcf; // EXTIN_* if (extout_mask == 0) - extout_mask = 0x7fff; + extout_mask = 0x7fff; // EXTOUT_* emu->fx8010.extin_mask = extin_mask; emu->fx8010.extout_mask = extout_mask; emu->enable_ir = enable_ir; @@ -1970,12 +1910,10 @@ int snd_emu10k1_create(struct snd_card *card, pgtbl[idx] = cpu_to_le32(silent_page | idx); /* set up voice indices */ - for (idx = 0; idx < NUM_G; idx++) { - emu->voices[idx].emu = emu; + for (idx = 0; idx < NUM_G; idx++) emu->voices[idx].number = idx; - } - err = snd_emu10k1_init(emu, enable_ir, 0); + err = snd_emu10k1_init(emu, enable_ir); if (err < 0) return err; #ifdef CONFIG_PM_SLEEP @@ -2007,7 +1945,7 @@ static const unsigned char saved_regs[] = { 0xff /* end */ }; static const unsigned char saved_regs_audigy[] = { - A_ADCIDX, A_MICIDX, A_FXWC1, A_FXWC2, A_SAMPLE_RATE, + A_ADCIDX, A_MICIDX, A_FXWC1, A_FXWC2, A_EHC, A_FXRT2, A_SENDAMOUNTS, A_FXRT1, 0xff /* end */ }; @@ -2054,7 +1992,7 @@ void snd_emu10k1_suspend_regs(struct snd_emu10k1 *emu) *val = snd_emu10k1_ptr_read(emu, *reg, i); } if (emu->audigy) - emu->saved_a_iocfg = inl(emu->port + A_IOCFG); + emu->saved_a_iocfg = inw(emu->port + A_IOCFG); emu->saved_hcfg = inl(emu->port + HCFG); } @@ -2068,7 +2006,7 @@ void snd_emu10k1_resume_init(struct snd_emu10k1 *emu) snd_emu10k1_emu1010_init(emu); else snd_emu10k1_ptr_write(emu, AC97SLOT, 0, AC97SLOT_CNTR|AC97SLOT_LFE); - snd_emu10k1_init(emu, emu->enable_ir, 1); + snd_emu10k1_init(emu, emu->enable_ir); } void snd_emu10k1_resume_regs(struct snd_emu10k1 *emu) @@ -2081,7 +2019,7 @@ void snd_emu10k1_resume_regs(struct snd_emu10k1 *emu) /* resore for spdif */ if (emu->audigy) - outl(emu->saved_a_iocfg, emu->port + A_IOCFG); + outw(emu->saved_a_iocfg, emu->port + A_IOCFG); outl(emu->saved_hcfg, emu->port + HCFG); val = emu->saved_ptr; diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index 6cf7c8b1de47..3f64ccab0e63 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -641,8 +641,8 @@ snd_emu10k1_look_for_ctl(struct snd_emu10k1 *emu, list_for_each_entry(ctl, &emu->fx8010.gpr_ctl, list) { kcontrol = ctl->kcontrol; if (kcontrol->id.iface == id->iface && - !strcmp(kcontrol->id.name, id->name) && - kcontrol->id.index == id->index) + kcontrol->id.index == id->index && + !strcmp(kcontrol->id.name, id->name)) return ctl; } return NULL; @@ -1187,8 +1187,8 @@ snd_emu10k1_init_stereo_onoff_control(struct snd_emu10k1_fx8010_control_gpr *ctl } /* - * Used for emu1010 - conversion from 32-bit capture inputs from HANA - * to 2 x 16-bit registers in audigy - their values are read via DMA. + * Used for emu1010 - conversion from 32-bit capture inputs from the FPGA + * to 2 x 16-bit registers in Audigy - their values are read via DMA. * Conversion is performed by Audigy DSP instructions of FX8010. */ static int snd_emu10k1_audigy_dsp_convert_32_to_2x16( @@ -1213,7 +1213,7 @@ static int snd_emu10k1_audigy_dsp_convert_32_to_2x16( static int _snd_emu10k1_audigy_init_efx(struct snd_emu10k1 *emu) { - int err, i, z, gpr, nctl; + int err, z, gpr, nctl; int bit_shifter16; const int playback = 10; const int capture = playback + (SND_EMU10K1_PLAYBACK_CHANNELS * 2); /* we reserve 10 voices */ @@ -1245,12 +1245,10 @@ static int _snd_emu10k1_audigy_init_efx(struct snd_emu10k1 *emu) icode->code = icode->tram_addr_map + 256; /* clear free GPRs */ - for (i = 0; i < 512; i++) - set_bit(i, icode->gpr_valid); + memset(icode->gpr_valid, 0xff, 512 / 8); /* clear TRAM data & address lines */ - for (i = 0; i < 256; i++) - set_bit(i, icode->tram_valid); + memset(icode->tram_valid, 0xff, 256 / 8); strcpy(icode->name, "Audigy DSP code for ALSA"); ptr = 0; @@ -1261,9 +1259,6 @@ static int _snd_emu10k1_audigy_init_efx(struct snd_emu10k1 *emu) gpr_map[gpr++] = 0x0000ffff; bit_shifter16 = gpr; - /* stop FX processor */ - snd_emu10k1_ptr_write(emu, A_DBG, 0, (emu->fx8010.dbg = 0) | A_DBG_SINGLE_STEP); - #if 1 /* PCM front Playback Volume (independent from stereo mix) * playback = 0 + ( gpr * FXBUS_PCM_LEFT_FRONT >> 31) @@ -1332,8 +1327,9 @@ static int _snd_emu10k1_audigy_init_efx(struct snd_emu10k1 *emu) #define A_ADD_VOLUME_IN(var,vol,input) \ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) - /* emu1212 DSP 0 and DSP 1 Capture */ if (emu->card_capabilities->emu_model) { + /* EMU1010 DSP 0 and DSP 1 Capture */ + // The 24 MSB hold the actual value. We implicitly discard the 16 LSB. if (emu->card_capabilities->ca0108_chip) { /* Note:JCD:No longer bit shift lower 16bits to upper 16bits of 32bit value. */ A_OP(icode, &ptr, iMACINT0, A_GPR(tmp), A_C_00000000, A3_EMU32IN(0x0), A_C_00000001); @@ -1359,7 +1355,7 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) gpr += 2; /* mic capture buffer */ - A_OP(icode, &ptr, iINTERP, A_EXTOUT(A_EXTOUT_MIC_CAP), A_EXTIN(A_EXTIN_AC97_L), 0xcd, A_EXTIN(A_EXTIN_AC97_R)); + A_OP(icode, &ptr, iINTERP, A_EXTOUT(A_EXTOUT_MIC_CAP), A_EXTIN(A_EXTIN_AC97_L), A_C_40000000, A_EXTIN(A_EXTIN_AC97_R)); /* Audigy CD Playback Volume */ A_ADD_VOLUME_IN(stereo_mix, gpr, A_EXTIN_SPDIF_CD_L); @@ -1442,7 +1438,7 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) /* Stereo Mix Center Playback */ /* Center = sub = Left/2 + Right/2 */ - A_OP(icode, &ptr, iINTERP, A_GPR(tmp), A_GPR(stereo_mix), 0xcd, A_GPR(stereo_mix+1)); + A_OP(icode, &ptr, iINTERP, A_GPR(tmp), A_GPR(stereo_mix), A_C_40000000, A_GPR(stereo_mix+1)); A_OP(icode, &ptr, iMAC0, A_GPR(playback+4), A_GPR(playback+4), A_GPR(gpr), A_GPR(tmp)); snd_emu10k1_init_mono_control(&controls[nctl++], "Center Playback Volume", gpr, 0); gpr++; @@ -1638,8 +1634,11 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) #endif if (emu->card_capabilities->emu_model) { + /* Capture 16 channels of S32_LE sound. */ if (emu->card_capabilities->ca0108_chip) { dev_info(emu->card->dev, "EMU2 inputs on\n"); + /* Note that the Tina[2] DSPs have 16 more EMU32 inputs which we don't use. */ + for (z = 0; z < 0x10; z++) { snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, @@ -1648,7 +1647,7 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) } } else { dev_info(emu->card->dev, "EMU inputs on\n"); - /* Capture 16 (originally 8) channels of S32_LE sound */ + /* Note that the Alice2 DSPs have 6 I2S inputs which we don't use. */ /* dev_dbg(emu->card->dev, "emufx.c: gpr=0x%x, tmp=0x%x\n", @@ -1886,12 +1885,10 @@ static int _snd_emu10k1_init_efx(struct snd_emu10k1 *emu) icode->code = icode->tram_addr_map + 160; /* clear free GPRs */ - for (i = 0; i < 256; i++) - set_bit(i, icode->gpr_valid); + memset(icode->gpr_valid, 0xff, 256 / 8); /* clear TRAM data & address lines */ - for (i = 0; i < 160; i++) - set_bit(i, icode->tram_valid); + memset(icode->tram_valid, 0xff, 160 / 8); strcpy(icode->name, "SB Live! FX8010 code for ALSA v1.2 by Jaroslav Kysela"); ptr = 0; i = 0; @@ -1903,9 +1900,6 @@ static int _snd_emu10k1_init_efx(struct snd_emu10k1 *emu) tmp = 0x88; /* we need 4 temporary GPR */ /* from 0x8c to 0xff is the area for tone control */ - /* stop FX processor */ - snd_emu10k1_ptr_write(emu, DBG, 0, (emu->fx8010.dbg = 0) | EMU10K1_DBG_SINGLE_STEP); - /* * Process FX Buses */ @@ -2484,7 +2478,7 @@ int snd_emu10k1_fx8010_tram_setup(struct snd_emu10k1 *emu, u32 size) outl(HCFG_LOCKTANKCACHE_MASK | inl(emu->port + HCFG), emu->port + HCFG); spin_unlock_irq(&emu->emu_lock); snd_emu10k1_ptr_write(emu, TCB, 0, 0); - snd_emu10k1_ptr_write(emu, TCBS, 0, 0); + snd_emu10k1_ptr_write(emu, TCBS, 0, TCBS_BUFFSIZE_16K); if (emu->fx8010.etram_pages.area != NULL) { snd_dma_free_pages(&emu->fx8010.etram_pages); emu->fx8010.etram_pages.area = NULL; @@ -2523,7 +2517,7 @@ static void snd_emu10k1_fx8010_info(struct snd_emu10k1 *emu, struct snd_emu10k1_fx8010_info *info) { const char * const *fxbus, * const *extin, * const *extout; - unsigned short fxbus_mask, extin_mask, extout_mask; + unsigned short extin_mask, extout_mask; int res; info->internal_tram_size = emu->fx8010.itram_size; @@ -2531,11 +2525,10 @@ static void snd_emu10k1_fx8010_info(struct snd_emu10k1 *emu, fxbus = fxbuses; extin = emu->audigy ? audigy_ins : creative_ins; extout = emu->audigy ? audigy_outs : creative_outs; - fxbus_mask = emu->fx8010.fxbus_mask; - extin_mask = emu->fx8010.extin_mask; - extout_mask = emu->fx8010.extout_mask; + extin_mask = emu->audigy ? ~0 : emu->fx8010.extin_mask; + extout_mask = emu->audigy ? ~0 : emu->fx8010.extout_mask; for (res = 0; res < 16; res++, fxbus++, extin++, extout++) { - copy_string(info->fxbus_names[res], fxbus_mask & (1 << res) ? *fxbus : NULL, "FXBUS", res); + copy_string(info->fxbus_names[res], *fxbus, "FXBUS", res); copy_string(info->extin_names[res], extin_mask & (1 << res) ? *extin : NULL, "Unused", res); copy_string(info->extout_names[res], extout_mask & (1 << res) ? *extout : NULL, "Unused", res); } @@ -2651,17 +2644,19 @@ static int snd_emu10k1_fx8010_ioctl(struct snd_hwdep * hw, struct file *file, un return -EPERM; if (get_user(addr, (unsigned int __user *)argp)) return -EFAULT; - if (addr > 0x1ff) - return -EINVAL; - if (emu->audigy) - snd_emu10k1_ptr_write(emu, A_DBG, 0, emu->fx8010.dbg |= A_DBG_SINGLE_STEP | addr); - else - snd_emu10k1_ptr_write(emu, DBG, 0, emu->fx8010.dbg |= EMU10K1_DBG_SINGLE_STEP | addr); - udelay(10); - if (emu->audigy) - snd_emu10k1_ptr_write(emu, A_DBG, 0, emu->fx8010.dbg |= A_DBG_SINGLE_STEP | A_DBG_STEP_ADDR | addr); - else - snd_emu10k1_ptr_write(emu, DBG, 0, emu->fx8010.dbg |= EMU10K1_DBG_SINGLE_STEP | EMU10K1_DBG_STEP | addr); + if (emu->audigy) { + if (addr > A_DBG_STEP_ADDR) + return -EINVAL; + snd_emu10k1_ptr_write(emu, A_DBG, 0, emu->fx8010.dbg |= A_DBG_SINGLE_STEP); + udelay(10); + snd_emu10k1_ptr_write(emu, A_DBG, 0, emu->fx8010.dbg | A_DBG_STEP | addr); + } else { + if (addr > EMU10K1_DBG_SINGLE_STEP_ADDR) + return -EINVAL; + snd_emu10k1_ptr_write(emu, DBG, 0, emu->fx8010.dbg |= EMU10K1_DBG_SINGLE_STEP); + udelay(10); + snd_emu10k1_ptr_write(emu, DBG, 0, emu->fx8010.dbg | EMU10K1_DBG_STEP | addr); + } return 0; case SNDRV_EMU10K1_IOCTL_DBG_READ: if (emu->audigy) diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index 3c115f8ab96c..3ebc7c36a444 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -346,7 +346,7 @@ static const unsigned int emu1616_output_dst[] = { }; /* - * Data destinations - HANA outputs going to Alice2 (audigy) for + * Data destinations - FPGA outputs going to Alice2 (Audigy) for * capture (EMU32 + I2S links) * Each destination has an enum mixer control to choose a data source */ @@ -367,6 +367,7 @@ static const unsigned int emu1010_input_dst[] = { EMU_DST_ALICE2_EMU32_D, EMU_DST_ALICE2_EMU32_E, EMU_DST_ALICE2_EMU32_F, + /* These exist only on rev1 EMU1010 cards. */ EMU_DST_ALICE_I2S0_LEFT, EMU_DST_ALICE_I2S0_RIGHT, EMU_DST_ALICE_I2S1_LEFT, @@ -708,7 +709,7 @@ static int snd_emu1010_internal_clock_put(struct snd_kcontrol *kcontrol, /* 44100 */ /* Mute all */ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_MUTE ); - /* Default fallback clock 48kHz */ + /* Default fallback clock 44.1kHz */ snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, EMU_HANA_DEFCLOCK_44_1K ); /* Word Clock source, Internal 44.1kHz x1 */ snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, @@ -826,8 +827,8 @@ static int snd_emu1010_optical_out_put(struct snd_kcontrol *kcontrol, change = (emu->emu1010.optical_out != val); if (change) { emu->emu1010.optical_out = val; - tmp = (emu->emu1010.optical_in ? EMU_HANA_OPTICAL_IN_ADAT : 0) | - (emu->emu1010.optical_out ? EMU_HANA_OPTICAL_OUT_ADAT : 0); + tmp = (emu->emu1010.optical_in ? EMU_HANA_OPTICAL_IN_ADAT : EMU_HANA_OPTICAL_IN_SPDIF) | + (emu->emu1010.optical_out ? EMU_HANA_OPTICAL_OUT_ADAT : EMU_HANA_OPTICAL_OUT_SPDIF); snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, tmp); } return change; @@ -877,8 +878,8 @@ static int snd_emu1010_optical_in_put(struct snd_kcontrol *kcontrol, change = (emu->emu1010.optical_in != val); if (change) { emu->emu1010.optical_in = val; - tmp = (emu->emu1010.optical_in ? EMU_HANA_OPTICAL_IN_ADAT : 0) | - (emu->emu1010.optical_out ? EMU_HANA_OPTICAL_OUT_ADAT : 0); + tmp = (emu->emu1010.optical_in ? EMU_HANA_OPTICAL_IN_ADAT : EMU_HANA_OPTICAL_IN_SPDIF) | + (emu->emu1010.optical_out ? EMU_HANA_OPTICAL_OUT_ADAT : EMU_HANA_OPTICAL_OUT_SPDIF); snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, tmp); } return change; @@ -924,7 +925,7 @@ static int snd_audigy_i2c_capture_source_put(struct snd_kcontrol *kcontrol, struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); unsigned int source_id; unsigned int ngain, ogain; - u32 gpio; + u16 gpio; int change = 0; unsigned long flags; u32 source; @@ -941,11 +942,11 @@ static int snd_audigy_i2c_capture_source_put(struct snd_kcontrol *kcontrol, if (change) { snd_emu10k1_i2c_write(emu, ADC_MUX, 0); /* Mute input */ spin_lock_irqsave(&emu->emu_lock, flags); - gpio = inl(emu->port + A_IOCFG); + gpio = inw(emu->port + A_IOCFG); if (source_id==0) - outl(gpio | 0x4, emu->port + A_IOCFG); + outw(gpio | 0x4, emu->port + A_IOCFG); else - outl(gpio & ~0x4, emu->port + A_IOCFG); + outw(gpio & ~0x4, emu->port + A_IOCFG); spin_unlock_irqrestore(&emu->emu_lock, flags); ngain = emu->i2c_capture_volume[source_id][0]; /* Left */ @@ -1005,7 +1006,7 @@ static int snd_audigy_i2c_volume_put(struct snd_kcontrol *kcontrol, { struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); unsigned int ogain; - unsigned int ngain; + unsigned int ngain0, ngain1; unsigned int source_id; int change = 0; @@ -1014,24 +1015,24 @@ static int snd_audigy_i2c_volume_put(struct snd_kcontrol *kcontrol, /* capture_source: uinfo->value.enumerated.items = 2 */ if (source_id >= 2) return -EINVAL; + ngain0 = ucontrol->value.integer.value[0]; + ngain1 = ucontrol->value.integer.value[1]; + if (ngain0 > 0xff) + return -EINVAL; + if (ngain1 > 0xff) + return -EINVAL; ogain = emu->i2c_capture_volume[source_id][0]; /* Left */ - ngain = ucontrol->value.integer.value[0]; - if (ngain > 0xff) - return 0; - if (ogain != ngain) { + if (ogain != ngain0) { if (emu->i2c_capture_source == source_id) - snd_emu10k1_i2c_write(emu, ADC_ATTEN_ADCL, ((ngain) & 0xff) ); - emu->i2c_capture_volume[source_id][0] = ngain; + snd_emu10k1_i2c_write(emu, ADC_ATTEN_ADCL, ngain0); + emu->i2c_capture_volume[source_id][0] = ngain0; change = 1; } ogain = emu->i2c_capture_volume[source_id][1]; /* Right */ - ngain = ucontrol->value.integer.value[1]; - if (ngain > 0xff) - return 0; - if (ogain != ngain) { + if (ogain != ngain1) { if (emu->i2c_capture_source == source_id) - snd_emu10k1_i2c_write(emu, ADC_ATTEN_ADCR, ((ngain) & 0xff)); - emu->i2c_capture_volume[source_id][1] = ngain; + snd_emu10k1_i2c_write(emu, ADC_ATTEN_ADCR, ngain1); + emu->i2c_capture_volume[source_id][1] = ngain1; change = 1; } @@ -1632,7 +1633,7 @@ static int snd_emu10k1_shared_spdif_get(struct snd_kcontrol *kcontrol, struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); if (emu->audigy) - ucontrol->value.integer.value[0] = inl(emu->port + A_IOCFG) & A_IOCFG_GPOUT0 ? 1 : 0; + ucontrol->value.integer.value[0] = inw(emu->port + A_IOCFG) & A_IOCFG_GPOUT0 ? 1 : 0; else ucontrol->value.integer.value[0] = inl(emu->port + HCFG) & HCFG_GPOUT0 ? 1 : 0; if (emu->card_capabilities->invert_shared_spdif) @@ -1657,13 +1658,13 @@ static int snd_emu10k1_shared_spdif_put(struct snd_kcontrol *kcontrol, if ( emu->card_capabilities->i2c_adc) { /* Do nothing for Audigy 2 ZS Notebook */ } else if (emu->audigy) { - reg = inl(emu->port + A_IOCFG); + reg = inw(emu->port + A_IOCFG); val = sw ? A_IOCFG_GPOUT0 : 0; change = (reg & A_IOCFG_GPOUT0) != val; if (change) { reg &= ~A_IOCFG_GPOUT0; reg |= val; - outl(reg | val, emu->port + A_IOCFG); + outw(reg | val, emu->port + A_IOCFG); } } reg = inl(emu->port + HCFG); diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index 48af77ae8020..e8d2f0f6fbb3 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -76,23 +76,6 @@ static void snd_emu10k1_pcm_efx_interrupt(struct snd_emu10k1 *emu, snd_pcm_period_elapsed(emu->pcm_capture_efx_substream); } -static snd_pcm_uframes_t snd_emu10k1_efx_playback_pointer(struct snd_pcm_substream *substream) -{ - struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_emu10k1_pcm *epcm = runtime->private_data; - unsigned int ptr; - - if (!epcm->running) - return 0; - ptr = snd_emu10k1_ptr_read(emu, CCCA, epcm->voices[0]->number) & 0x00ffffff; - ptr += runtime->buffer_size; - ptr -= epcm->ccca_start_addr; - ptr %= runtime->buffer_size; - - return ptr; -} - static int snd_emu10k1_pcm_channel_alloc(struct snd_emu10k1_pcm * epcm, int voices) { int err, i; @@ -343,7 +326,6 @@ static void snd_emu10k1_pcm_init_voice(struct snd_emu10k1 *emu, } else snd_emu10k1_ptr_write(emu, FXRT, voice, snd_emu10k1_compose_send_routing(send_routing)); - /* Stop CA */ /* Assumption that PT is already 0 so no harm overwriting */ snd_emu10k1_ptr_write(emu, PTRX, voice, (send_amount[0] << 8) | send_amount[1]); snd_emu10k1_ptr_write(emu, DSL, voice, end_addr | (send_amount[3] << 24)); @@ -370,8 +352,8 @@ static void snd_emu10k1_pcm_init_voice(struct snd_emu10k1 *emu, snd_emu10k1_ptr_write(emu, MAPA, voice, silent_page); snd_emu10k1_ptr_write(emu, MAPB, voice, silent_page); /* modulation envelope */ - snd_emu10k1_ptr_write(emu, CVCF, voice, 0xffff); - snd_emu10k1_ptr_write(emu, VTFT, voice, 0xffff); + snd_emu10k1_ptr_write(emu, VTFT, voice, VTFT_FILTERTARGET_MASK); + snd_emu10k1_ptr_write(emu, CVCF, voice, CVCF_CURRENTFILTER_MASK); snd_emu10k1_ptr_write(emu, ATKHLDM, voice, 0); snd_emu10k1_ptr_write(emu, DCYSUSM, voice, 0x007f); snd_emu10k1_ptr_write(emu, LFOVAL1, voice, 0x8000); @@ -433,36 +415,6 @@ static int snd_emu10k1_playback_hw_free(struct snd_pcm_substream *substream) struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct snd_emu10k1_pcm *epcm; - - if (runtime->private_data == NULL) - return 0; - epcm = runtime->private_data; - if (epcm->extra) { - snd_emu10k1_voice_free(epcm->emu, epcm->extra); - epcm->extra = NULL; - } - if (epcm->voices[1]) { - snd_emu10k1_voice_free(epcm->emu, epcm->voices[1]); - epcm->voices[1] = NULL; - } - if (epcm->voices[0]) { - snd_emu10k1_voice_free(epcm->emu, epcm->voices[0]); - epcm->voices[0] = NULL; - } - if (epcm->memblk) { - snd_emu10k1_free_pages(emu, epcm->memblk); - epcm->memblk = NULL; - epcm->start_addr = 0; - } - snd_pcm_lib_free_pages(substream); - return 0; -} - -static int snd_emu10k1_efx_playback_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_emu10k1_pcm *epcm; int i; if (runtime->private_data == NULL) @@ -527,9 +479,6 @@ static int snd_emu10k1_efx_playback_prepare(struct snd_pcm_substream *substream) start_addr = epcm->start_addr; end_addr = epcm->start_addr + snd_pcm_lib_buffer_bytes(substream); - /* - * the kX driver leaves some space between voices - */ channel_size = ( end_addr - start_addr ) / NUM_EFX_PLAYBACK; snd_emu10k1_pcm_init_voice(emu, 1, 1, epcm->extra, @@ -672,8 +621,8 @@ static void snd_emu10k1_playback_prepare_voice(struct snd_emu10k1 *emu, struct s tmp = runtime->channels == 2 ? (master ? 1 : 2) : 0; vattn = mix != NULL ? (mix->attn[tmp] << 16) : 0; snd_emu10k1_ptr_write(emu, IFATN, voice, attn); - snd_emu10k1_ptr_write(emu, VTFT, voice, vattn | 0xffff); - snd_emu10k1_ptr_write(emu, CVCF, voice, vattn | 0xffff); + snd_emu10k1_ptr_write(emu, VTFT, voice, vattn | VTFT_FILTERTARGET_MASK); + snd_emu10k1_ptr_write(emu, CVCF, voice, vattn | CVCF_CURRENTFILTER_MASK); snd_emu10k1_ptr_write(emu, DCYSUSV, voice, 0x7f7f); snd_emu10k1_voice_clear_loop_stop(emu, voice); } @@ -714,8 +663,8 @@ static void snd_emu10k1_playback_stop_voice(struct snd_emu10k1 *emu, struct snd_ snd_emu10k1_ptr_write(emu, PTRX_PITCHTARGET, voice, 0); snd_emu10k1_ptr_write(emu, CPF_CURRENTPITCH, voice, 0); snd_emu10k1_ptr_write(emu, IFATN, voice, 0xffff); - snd_emu10k1_ptr_write(emu, VTFT, voice, 0xffff); - snd_emu10k1_ptr_write(emu, CVCF, voice, 0xffff); + snd_emu10k1_ptr_write(emu, VTFT, voice, VTFT_FILTERTARGET_MASK); + snd_emu10k1_ptr_write(emu, CVCF, voice, CVCF_CURRENTFILTER_MASK); snd_emu10k1_ptr_write(emu, IP, voice, 0); } @@ -1091,8 +1040,6 @@ static int snd_emu10k1_efx_playback_open(struct snd_pcm_substream *substream) epcm->type = PLAYBACK_EFX; epcm->substream = substream; - emu->pcm_playback_efx_substream = substream; - runtime->private_data = epcm; runtime->private_free = snd_emu10k1_pcm_free_substream; runtime->hw = snd_emu10k1_efx_playback; @@ -1236,7 +1183,7 @@ static int snd_emu10k1_capture_mic_close(struct snd_pcm_substream *substream) { struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream); - emu->capture_interrupt = NULL; + emu->capture_mic_interrupt = NULL; emu->pcm_capture_mic_substream = NULL; return 0; } @@ -1267,9 +1214,7 @@ static int snd_emu10k1_capture_efx_open(struct snd_pcm_substream *substream) runtime->hw.rate_min = runtime->hw.rate_max = 48000; spin_lock_irq(&emu->reg_lock); if (emu->card_capabilities->emu_model) { - /* Nb. of channels has been increased to 16 */ /* TODO - * SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE * SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | * SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | * SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000 @@ -1280,13 +1225,14 @@ static int snd_emu10k1_capture_efx_open(struct snd_pcm_substream *substream) * Need to add mixer control to fix sample rate * * There are 32 mono channels of 16bits each. - * 24bit Audio uses 2x channels over 16bit - * 96kHz uses 2x channels over 48kHz - * 192kHz uses 4x channels over 48kHz - * So, for 48kHz 24bit, one has 16 channels - * for 96kHz 24bit, one has 8 channels - * for 192kHz 24bit, one has 4 channels - * + * 24bit Audio uses 2x channels over 16bit, + * 96kHz uses 2x channels over 48kHz, + * 192kHz uses 4x channels over 48kHz. + * So, for 48kHz 24bit, one has 16 channels, + * for 96kHz 24bit, one has 8 channels, + * for 192kHz 24bit, one has 4 channels. + * 1010rev2 and 1616(m) cards have double that, + * but we don't exceed 16 channels anyway. */ #if 1 switch (emu->emu1010.internal_clock) { @@ -1344,7 +1290,7 @@ static int snd_emu10k1_capture_efx_close(struct snd_pcm_substream *substream) { struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream); - emu->capture_interrupt = NULL; + emu->capture_efx_interrupt = NULL; emu->pcm_capture_efx_substream = NULL; return 0; } @@ -1372,10 +1318,10 @@ static const struct snd_pcm_ops snd_emu10k1_efx_playback_ops = { .open = snd_emu10k1_efx_playback_open, .close = snd_emu10k1_efx_playback_close, .hw_params = snd_emu10k1_playback_hw_params, - .hw_free = snd_emu10k1_efx_playback_hw_free, + .hw_free = snd_emu10k1_playback_hw_free, .prepare = snd_emu10k1_efx_playback_prepare, .trigger = snd_emu10k1_efx_playback_trigger, - .pointer = snd_emu10k1_efx_playback_pointer, + .pointer = snd_emu10k1_playback_pointer, }; int snd_emu10k1_pcm(struct snd_emu10k1 *emu, int device) @@ -1508,11 +1454,12 @@ static int snd_emu10k1_pcm_efx_voices_mask_put(struct snd_kcontrol *kcontrol, st nval[idx / 32] |= 1 << (idx % 32); bits++; } - + + // Check that the number of requested channels is a power of two + // not bigger than the number of available channels. for (idx = 0; idx < nefxb; idx++) if (1 << idx == bits) break; - if (idx >= nefxb) return -EINVAL; @@ -1781,24 +1728,27 @@ int snd_emu10k1_pcm_efx(struct snd_emu10k1 *emu, int device) struct snd_kcontrol *kctl; int err; - err = snd_pcm_new(emu->card, "emu10k1 efx", device, 8, 1, &pcm); + err = snd_pcm_new(emu->card, "emu10k1 efx", device, emu->audigy ? 0 : 8, 1, &pcm); if (err < 0) return err; pcm->private_data = emu; - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_emu10k1_fx8010_playback_ops); + if (!emu->audigy) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_emu10k1_fx8010_playback_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_emu10k1_capture_efx_ops); pcm->info_flags = 0; - strcpy(pcm->name, "Multichannel Capture/PT Playback"); + if (emu->audigy) + strcpy(pcm->name, "Multichannel Capture"); + else + strcpy(pcm->name, "Multichannel Capture/PT Playback"); emu->pcm_efx = pcm; /* EFX capture - record the "FXBUS2" channels, by default we connect the EXTINs * to these */ - /* emu->efx_voices_mask[0] = FXWC_DEFAULTROUTE_C | FXWC_DEFAULTROUTE_A; */ if (emu->audigy) { emu->efx_voices_mask[0] = 0; if (emu->card_capabilities->emu_model) diff --git a/sound/pci/emu10k1/emuproc.c b/sound/pci/emu10k1/emuproc.c index 6e20cca9c98f..bec72dc60a41 100644 --- a/sound/pci/emu10k1/emuproc.c +++ b/sound/pci/emu10k1/emuproc.c @@ -477,10 +477,7 @@ static void snd_emu_proc_ptr_reg_read(struct snd_info_entry *entry, for(i = offset; i < offset+length; i++) { snd_iprintf(buffer, "%02X: ",i); for (j = 0; j < voices; j++) { - if(iobase == 0) - value = snd_ptr_read(emu, 0, i, j); - else - value = snd_ptr_read(emu, 0x20, i, j); + value = snd_ptr_read(emu, iobase, i, j); snd_iprintf(buffer, "%08lX ", value); } snd_iprintf(buffer, "\n"); diff --git a/sound/pci/emu10k1/io.c b/sound/pci/emu10k1/io.c index e15092ce9848..cfb96a67aa35 100644 --- a/sound/pci/emu10k1/io.c +++ b/sound/pci/emu10k1/io.c @@ -95,8 +95,8 @@ unsigned int snd_emu10k1_ptr20_read(struct snd_emu10k1 * emu, regptr = (reg << 16) | chn; spin_lock_irqsave(&emu->emu_lock, flags); - outl(regptr, emu->port + 0x20 + PTR); - val = inl(emu->port + 0x20 + DATA); + outl(regptr, emu->port + PTR2); + val = inl(emu->port + DATA2); spin_unlock_irqrestore(&emu->emu_lock, flags); return val; } @@ -112,8 +112,8 @@ void snd_emu10k1_ptr20_write(struct snd_emu10k1 *emu, regptr = (reg << 16) | chn; spin_lock_irqsave(&emu->emu_lock, flags); - outl(regptr, emu->port + 0x20 + PTR); - outl(data, emu->port + 0x20 + DATA); + outl(regptr, emu->port + PTR2); + outl(data, emu->port + DATA2); spin_unlock_irqrestore(&emu->emu_lock, flags); } @@ -128,7 +128,7 @@ int snd_emu10k1_spi_write(struct snd_emu10k1 * emu, /* This function is not re-entrant, so protect against it. */ spin_lock(&emu->spi_lock); if (emu->card_capabilities->ca0108_chip) - reg = 0x3c; /* PTR20, reg 0x3c */ + reg = P17V_SPI; else { /* For other chip types the SPI register * is currently unknown. */ @@ -233,56 +233,57 @@ int snd_emu10k1_i2c_write(struct snd_emu10k1 *emu, return err; } -int snd_emu1010_fpga_write(struct snd_emu10k1 * emu, u32 reg, u32 value) +void snd_emu1010_fpga_write(struct snd_emu10k1 *emu, u32 reg, u32 value) { unsigned long flags; - if (reg > 0x3f) - return 1; + if (snd_BUG_ON(reg > 0x3f)) + return; reg += 0x40; /* 0x40 upwards are registers. */ - if (value > 0x3f) /* 0 to 0x3f are values */ - return 1; + if (snd_BUG_ON(value > 0x3f)) /* 0 to 0x3f are values */ + return; spin_lock_irqsave(&emu->emu_lock, flags); - outl(reg, emu->port + A_IOCFG); + outw(reg, emu->port + A_GPIO); udelay(10); - outl(reg | 0x80, emu->port + A_IOCFG); /* High bit clocks the value into the fpga. */ + outw(reg | 0x80, emu->port + A_GPIO); /* High bit clocks the value into the fpga. */ udelay(10); - outl(value, emu->port + A_IOCFG); + outw(value, emu->port + A_GPIO); udelay(10); - outl(value | 0x80 , emu->port + A_IOCFG); /* High bit clocks the value into the fpga. */ + outw(value | 0x80 , emu->port + A_GPIO); /* High bit clocks the value into the fpga. */ spin_unlock_irqrestore(&emu->emu_lock, flags); - - return 0; } -int snd_emu1010_fpga_read(struct snd_emu10k1 * emu, u32 reg, u32 *value) +void snd_emu1010_fpga_read(struct snd_emu10k1 *emu, u32 reg, u32 *value) { + // The higest input pin is used as the designated interrupt trigger, + // so it needs to be masked out. + u32 mask = emu->card_capabilities->ca0108_chip ? 0x1f : 0x7f; unsigned long flags; - if (reg > 0x3f) - return 1; + if (snd_BUG_ON(reg > 0x3f)) + return; reg += 0x40; /* 0x40 upwards are registers. */ spin_lock_irqsave(&emu->emu_lock, flags); - outl(reg, emu->port + A_IOCFG); + outw(reg, emu->port + A_GPIO); udelay(10); - outl(reg | 0x80, emu->port + A_IOCFG); /* High bit clocks the value into the fpga. */ + outw(reg | 0x80, emu->port + A_GPIO); /* High bit clocks the value into the fpga. */ udelay(10); - *value = ((inl(emu->port + A_IOCFG) >> 8) & 0x7f); + *value = ((inw(emu->port + A_GPIO) >> 8) & mask); spin_unlock_irqrestore(&emu->emu_lock, flags); - - return 0; } /* Each Destination has one and only one Source, * but one Source can feed any number of Destinations simultaneously. */ -int snd_emu1010_fpga_link_dst_src_write(struct snd_emu10k1 * emu, u32 dst, u32 src) +void snd_emu1010_fpga_link_dst_src_write(struct snd_emu10k1 *emu, u32 dst, u32 src) { - snd_emu1010_fpga_write(emu, 0x00, ((dst >> 8) & 0x3f) ); - snd_emu1010_fpga_write(emu, 0x01, (dst & 0x3f) ); - snd_emu1010_fpga_write(emu, 0x02, ((src >> 8) & 0x3f) ); - snd_emu1010_fpga_write(emu, 0x03, (src & 0x3f) ); - - return 0; + if (snd_BUG_ON(dst & ~0x71f)) + return; + if (snd_BUG_ON(src & ~0x71f)) + return; + snd_emu1010_fpga_write(emu, EMU_HANA_DESTHI, dst >> 8); + snd_emu1010_fpga_write(emu, EMU_HANA_DESTLO, dst & 0x1f); + snd_emu1010_fpga_write(emu, EMU_HANA_SRCHI, src >> 8); + snd_emu1010_fpga_write(emu, EMU_HANA_SRCLO, src & 0x1f); } void snd_emu10k1_intr_enable(struct snd_emu10k1 *emu, unsigned int intrenb) @@ -313,7 +314,6 @@ void snd_emu10k1_voice_intr_enable(struct snd_emu10k1 *emu, unsigned int voicenu unsigned int val; spin_lock_irqsave(&emu->emu_lock, flags); - /* voice interrupt */ if (voicenum >= 32) { outl(CLIEH << 16, emu->port + PTR); val = inl(emu->port + DATA); @@ -333,7 +333,6 @@ void snd_emu10k1_voice_intr_disable(struct snd_emu10k1 *emu, unsigned int voicen unsigned int val; spin_lock_irqsave(&emu->emu_lock, flags); - /* voice interrupt */ if (voicenum >= 32) { outl(CLIEH << 16, emu->port + PTR); val = inl(emu->port + DATA); @@ -352,7 +351,6 @@ void snd_emu10k1_voice_intr_ack(struct snd_emu10k1 *emu, unsigned int voicenum) unsigned long flags; spin_lock_irqsave(&emu->emu_lock, flags); - /* voice interrupt */ if (voicenum >= 32) { outl(CLIPH << 16, emu->port + PTR); voicenum = 1 << (voicenum - 32); @@ -370,7 +368,6 @@ void snd_emu10k1_voice_half_loop_intr_enable(struct snd_emu10k1 *emu, unsigned i unsigned int val; spin_lock_irqsave(&emu->emu_lock, flags); - /* voice interrupt */ if (voicenum >= 32) { outl(HLIEH << 16, emu->port + PTR); val = inl(emu->port + DATA); @@ -390,7 +387,6 @@ void snd_emu10k1_voice_half_loop_intr_disable(struct snd_emu10k1 *emu, unsigned unsigned int val; spin_lock_irqsave(&emu->emu_lock, flags); - /* voice interrupt */ if (voicenum >= 32) { outl(HLIEH << 16, emu->port + PTR); val = inl(emu->port + DATA); @@ -409,7 +405,6 @@ void snd_emu10k1_voice_half_loop_intr_ack(struct snd_emu10k1 *emu, unsigned int unsigned long flags; spin_lock_irqsave(&emu->emu_lock, flags); - /* voice interrupt */ if (voicenum >= 32) { outl(HLIPH << 16, emu->port + PTR); voicenum = 1 << (voicenum - 32); @@ -427,7 +422,6 @@ void snd_emu10k1_voice_set_loop_stop(struct snd_emu10k1 *emu, unsigned int voice unsigned int sol; spin_lock_irqsave(&emu->emu_lock, flags); - /* voice interrupt */ if (voicenum >= 32) { outl(SOLEH << 16, emu->port + PTR); sol = inl(emu->port + DATA); @@ -447,7 +441,6 @@ void snd_emu10k1_voice_clear_loop_stop(struct snd_emu10k1 *emu, unsigned int voi unsigned int sol; spin_lock_irqsave(&emu->emu_lock, flags); - /* voice interrupt */ if (voicenum >= 32) { outl(SOLEH << 16, emu->port + PTR); sol = inl(emu->port + DATA); diff --git a/sound/pci/emu10k1/irq.c b/sound/pci/emu10k1/irq.c index ebb2275efb6c..dfb44e5e69a7 100644 --- a/sound/pci/emu10k1/irq.c +++ b/sound/pci/emu10k1/irq.c @@ -18,7 +18,7 @@ irqreturn_t snd_emu10k1_interrupt(int irq, void *dev_id) { struct snd_emu10k1 *emu = dev_id; - unsigned int status, status2, orig_status, orig_status2; + unsigned int status, orig_status; int handled = 0; int timeout = 0; @@ -139,32 +139,10 @@ irqreturn_t snd_emu10k1_interrupt(int irq, void *dev_id) status &= ~IPR_FXDSP; } if (status & IPR_P16V) { - while ((status2 = inl(emu->port + IPR2)) != 0) { - u32 mask = INTE2_PLAYBACK_CH_0_LOOP; /* Full Loop */ - struct snd_emu10k1_voice *pvoice = &(emu->p16v_voices[0]); - struct snd_emu10k1_voice *cvoice = &(emu->p16v_capture_voice); - - /* dev_dbg(emu->card->dev, "status2=0x%x\n", status2); */ - orig_status2 = status2; - if(status2 & mask) { - if(pvoice->use) { - snd_pcm_period_elapsed(pvoice->epcm->substream); - } else { - dev_err(emu->card->dev, - "p16v: status: 0x%08x, mask=0x%08x, pvoice=%p, use=%d\n", - status2, mask, pvoice, - pvoice->use); - } - } - if(status2 & 0x110000) { - /* dev_info(emu->card->dev, "capture int found\n"); */ - if(cvoice->use) { - /* dev_info(emu->card->dev, "capture period_elapsed\n"); */ - snd_pcm_period_elapsed(cvoice->epcm->substream); - } - } - outl(orig_status2, emu->port + IPR2); /* ack all */ - } + if (emu->p16v_interrupt) + emu->p16v_interrupt(emu); + else + outl(0, emu->port + INTE2); status &= ~IPR_P16V; } diff --git a/sound/pci/emu10k1/p16v.c b/sound/pci/emu10k1/p16v.c index 18a1b0740e6b..e7f097cae574 100644 --- a/sound/pci/emu10k1/p16v.c +++ b/sound/pci/emu10k1/p16v.c @@ -149,42 +149,19 @@ static const struct snd_pcm_hardware snd_p16v_capture_hw = { .fifo_size = 0, }; -static void snd_p16v_pcm_free_substream(struct snd_pcm_runtime *runtime) -{ - struct snd_emu10k1_pcm *epcm = runtime->private_data; - - kfree(epcm); -} - /* open_playback callback */ static int snd_p16v_pcm_open_playback_channel(struct snd_pcm_substream *substream, int channel_id) { - struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream); - struct snd_emu10k1_voice *channel = &(emu->p16v_voices[channel_id]); - struct snd_emu10k1_pcm *epcm; struct snd_pcm_runtime *runtime = substream->runtime; int err; - epcm = kzalloc(sizeof(*epcm), GFP_KERNEL); - /* dev_dbg(emu->card->dev, "epcm kcalloc: %p\n", epcm); */ - - if (epcm == NULL) - return -ENOMEM; - epcm->emu = emu; - epcm->substream = substream; /* dev_dbg(emu->card->dev, "epcm device=%d, channel_id=%d\n", substream->pcm->device, channel_id); */ - runtime->private_data = epcm; - runtime->private_free = snd_p16v_pcm_free_substream; runtime->hw = snd_p16v_playback_hw; - channel->emu = emu; - channel->number = channel_id; - - channel->use=1; #if 0 /* debug */ dev_dbg(emu->card->dev, "p16v: open channel_id=%d, channel=%p, use=0x%x\n", @@ -193,7 +170,6 @@ static int snd_p16v_pcm_open_playback_channel(struct snd_pcm_substream *substrea channel_id, chip, channel); #endif /* debug */ /* channel->interrupt = snd_p16v_pcm_channel_interrupt; */ - channel->epcm = epcm; err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); if (err < 0) return err; @@ -205,44 +181,20 @@ static int snd_p16v_pcm_open_playback_channel(struct snd_pcm_substream *substrea return 0; } + /* open_capture callback */ static int snd_p16v_pcm_open_capture_channel(struct snd_pcm_substream *substream, int channel_id) { - struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream); - struct snd_emu10k1_voice *channel = &(emu->p16v_capture_voice); - struct snd_emu10k1_pcm *epcm; struct snd_pcm_runtime *runtime = substream->runtime; int err; - epcm = kzalloc(sizeof(*epcm), GFP_KERNEL); - /* dev_dbg(emu->card->dev, "epcm kcalloc: %p\n", epcm); */ - - if (epcm == NULL) - return -ENOMEM; - epcm->emu = emu; - epcm->substream = substream; /* dev_dbg(emu->card->dev, "epcm device=%d, channel_id=%d\n", substream->pcm->device, channel_id); */ - runtime->private_data = epcm; - runtime->private_free = snd_p16v_pcm_free_substream; runtime->hw = snd_p16v_capture_hw; - channel->emu = emu; - channel->number = channel_id; - - channel->use=1; -#if 0 /* debug */ - dev_dbg(emu->card->dev, - "p16v: open channel_id=%d, channel=%p, use=0x%x\n", - channel_id, channel, channel->use); - dev_dbg(emu->card->dev, "open:channel_id=%d, chip=%p, channel=%p\n", - channel_id, chip, channel); -#endif /* debug */ - /* channel->interrupt = snd_p16v_pcm_channel_interrupt; */ - channel->epcm = epcm; err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); if (err < 0) return err; @@ -254,22 +206,12 @@ static int snd_p16v_pcm_open_capture_channel(struct snd_pcm_substream *substream /* close callback */ static int snd_p16v_pcm_close_playback(struct snd_pcm_substream *substream) { - struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream); - //struct snd_pcm_runtime *runtime = substream->runtime; - //struct snd_emu10k1_pcm *epcm = runtime->private_data; - emu->p16v_voices[substream->pcm->device - emu->p16v_device_offset].use = 0; - /* FIXME: maybe zero others */ return 0; } /* close callback */ static int snd_p16v_pcm_close_capture(struct snd_pcm_substream *substream) { - struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream); - //struct snd_pcm_runtime *runtime = substream->runtime; - //struct snd_emu10k1_pcm *epcm = runtime->private_data; - emu->p16v_capture_voice.use = 0; - /* FIXME: maybe zero others */ return 0; } @@ -312,19 +254,24 @@ static int snd_p16v_pcm_prepare_playback(struct snd_pcm_substream *substream) emu->p16v_buffer->bytes); #endif /* debug */ tmp = snd_emu10k1_ptr_read(emu, A_SPDIF_SAMPLERATE, channel); + tmp &= ~(A_SPDIF_RATE_MASK | A_EHC_SRC48_MASK); switch (runtime->rate) { case 44100: - snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, channel, (tmp & ~0xe0e0) | 0x8080); + snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, channel, + tmp | A_SPDIF_44100 | A_EHC_SRC48_44); break; case 96000: - snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, channel, (tmp & ~0xe0e0) | 0x4040); + snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, channel, + tmp | A_SPDIF_96000 | A_EHC_SRC48_96); break; case 192000: - snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, channel, (tmp & ~0xe0e0) | 0x2020); + snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, channel, + tmp | A_SPDIF_192000 | A_EHC_SRC48_192); break; case 48000: default: - snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, channel, (tmp & ~0xe0e0) | 0x0000); + snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, channel, + tmp | A_SPDIF_48000 | A_EHC_SRC48_BYPASS); break; } /* FIXME: Check emu->buffer.size before actually writing to it. */ @@ -340,8 +287,8 @@ static int snd_p16v_pcm_prepare_playback(struct snd_pcm_substream *substream) //snd_emu10k1_ptr20_write(emu, PLAYBACK_PERIOD_SIZE, channel, frames_to_bytes(runtime, runtime->period_size)<<16); // buffer size in bytes snd_emu10k1_ptr20_write(emu, PLAYBACK_PERIOD_SIZE, channel, 0); // buffer size in bytes snd_emu10k1_ptr20_write(emu, PLAYBACK_POINTER, channel, 0); - snd_emu10k1_ptr20_write(emu, 0x07, channel, 0x0); - snd_emu10k1_ptr20_write(emu, 0x08, channel, 0); + snd_emu10k1_ptr20_write(emu, PLAYBACK_FIFO_END_ADDRESS, channel, 0); + snd_emu10k1_ptr20_write(emu, PLAYBACK_FIFO_POINTER, channel, 0); return 0; } @@ -352,7 +299,6 @@ static int snd_p16v_pcm_prepare_capture(struct snd_pcm_substream *substream) struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; int channel = substream->pcm->device - emu->p16v_device_offset; - u32 tmp; /* dev_dbg(emu->card->dev, "prepare capture:channel_number=%d, rate=%d, " @@ -362,24 +308,23 @@ static int snd_p16v_pcm_prepare_capture(struct snd_pcm_substream *substream) runtime->buffer_size, runtime->period_size, frames_to_bytes(runtime, 1)); */ - tmp = snd_emu10k1_ptr_read(emu, A_SPDIF_SAMPLERATE, channel); switch (runtime->rate) { case 44100: - snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, channel, (tmp & ~0x0e00) | 0x0800); + snd_emu10k1_ptr_write(emu, A_I2S_CAPTURE_RATE, channel, A_I2S_CAPTURE_44100); break; case 96000: - snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, channel, (tmp & ~0x0e00) | 0x0400); + snd_emu10k1_ptr_write(emu, A_I2S_CAPTURE_RATE, channel, A_I2S_CAPTURE_96000); break; case 192000: - snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, channel, (tmp & ~0x0e00) | 0x0200); + snd_emu10k1_ptr_write(emu, A_I2S_CAPTURE_RATE, channel, A_I2S_CAPTURE_192000); break; case 48000: default: - snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, channel, (tmp & ~0x0e00) | 0x0000); + snd_emu10k1_ptr_write(emu, A_I2S_CAPTURE_RATE, channel, A_I2S_CAPTURE_48000); break; } /* FIXME: Check emu->buffer.size before actually writing to it. */ - snd_emu10k1_ptr20_write(emu, 0x13, channel, 0); + snd_emu10k1_ptr20_write(emu, CAPTURE_FIFO_POINTER, channel, 0); snd_emu10k1_ptr20_write(emu, CAPTURE_DMA_ADDR, channel, runtime->dma_addr); snd_emu10k1_ptr20_write(emu, CAPTURE_BUFFER_SIZE, channel, frames_to_bytes(runtime, runtime->buffer_size) << 16); // buffer size in bytes snd_emu10k1_ptr20_write(emu, CAPTURE_POINTER, channel, 0); @@ -411,13 +356,48 @@ static void snd_p16v_intr_disable(struct snd_emu10k1 *emu, unsigned int intrenb) spin_unlock_irqrestore(&emu->emu_lock, flags); } +static void snd_p16v_interrupt(struct snd_emu10k1 *emu) +{ + unsigned int status; + + while ((status = inl(emu->port + IPR2)) != 0) { + u32 mask = INTE2_PLAYBACK_CH_0_LOOP; /* Full Loop */ + + /* dev_dbg(emu->card->dev, "p16v status=0x%x\n", status); */ + if (status & mask) { + struct snd_pcm_substream *substream = + emu->pcm_p16v->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + struct snd_pcm_runtime *runtime = substream->runtime; + + if (runtime && runtime->private_data) { + snd_pcm_period_elapsed(substream); + } else { + dev_err(emu->card->dev, + "p16v: status: 0x%08x, mask=0x%08x\n", + status, mask); + } + } + if (status & 0x110000) { + struct snd_pcm_substream *substream = + emu->pcm_p16v->streams[SNDRV_PCM_STREAM_CAPTURE].substream; + struct snd_pcm_runtime *runtime = substream->runtime; + + /* dev_info(emu->card->dev, "capture int found\n"); */ + if (runtime && runtime->private_data) { + /* dev_info(emu->card->dev, "capture period_elapsed\n"); */ + snd_pcm_period_elapsed(substream); + } + } + outl(status, emu->port + IPR2); /* ack all */ + } +} + /* trigger_playback callback */ static int snd_p16v_pcm_trigger_playback(struct snd_pcm_substream *substream, int cmd) { struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime; - struct snd_emu10k1_pcm *epcm; int channel; int result = 0; struct snd_pcm_substream *s; @@ -439,10 +419,9 @@ static int snd_p16v_pcm_trigger_playback(struct snd_pcm_substream *substream, s->stream != SNDRV_PCM_STREAM_PLAYBACK) continue; runtime = s->runtime; - epcm = runtime->private_data; channel = substream->pcm->device-emu->p16v_device_offset; /* dev_dbg(emu->card->dev, "p16v channel=%d\n", channel); */ - epcm->running = running; + runtime->private_data = (void *)(ptrdiff_t)running; basic |= (0x1<<channel); inte |= (INTE2_PLAYBACK_CH_0_LOOP<<channel); snd_pcm_trigger_done(s, substream); @@ -471,7 +450,6 @@ static int snd_p16v_pcm_trigger_capture(struct snd_pcm_substream *substream, { struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_emu10k1_pcm *epcm = runtime->private_data; int channel = 0; int result = 0; u32 inte = INTE2_CAPTURE_CH_0_LOOP | INTE2_CAPTURE_CH_0_HALF_LOOP; @@ -480,13 +458,13 @@ static int snd_p16v_pcm_trigger_capture(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: snd_p16v_intr_enable(emu, inte); snd_emu10k1_ptr20_write(emu, BASIC_INTERRUPT, 0, snd_emu10k1_ptr20_read(emu, BASIC_INTERRUPT, 0)|(0x100<<channel)); - epcm->running = 1; + runtime->private_data = (void *)1; break; case SNDRV_PCM_TRIGGER_STOP: snd_emu10k1_ptr20_write(emu, BASIC_INTERRUPT, 0, snd_emu10k1_ptr20_read(emu, BASIC_INTERRUPT, 0) & ~(0x100<<channel)); snd_p16v_intr_disable(emu, inte); //snd_emu10k1_ptr20_write(emu, EXTENDED_INT_MASK, 0, snd_emu10k1_ptr20_read(emu, EXTENDED_INT_MASK, 0) & ~(0x110000<<channel)); - epcm->running = 0; + runtime->private_data = NULL; break; default: result = -EINVAL; @@ -501,10 +479,10 @@ snd_p16v_pcm_pointer_playback(struct snd_pcm_substream *substream) { struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_emu10k1_pcm *epcm = runtime->private_data; snd_pcm_uframes_t ptr, ptr1, ptr2,ptr3,ptr4 = 0; int channel = substream->pcm->device - emu->p16v_device_offset; - if (!epcm->running) + + if (!runtime->private_data) return 0; ptr3 = snd_emu10k1_ptr20_read(emu, PLAYBACK_LIST_PTR, channel); @@ -526,11 +504,10 @@ snd_p16v_pcm_pointer_capture(struct snd_pcm_substream *substream) { struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_emu10k1_pcm *epcm = runtime->private_data; snd_pcm_uframes_t ptr, ptr1, ptr2 = 0; int channel = 0; - if (!epcm->running) + if (!runtime->private_data) return 0; ptr1 = snd_emu10k1_ptr20_read(emu, CAPTURE_POINTER, channel); @@ -591,6 +568,7 @@ int snd_p16v_pcm(struct snd_emu10k1 *emu, int device) pcm->dev_subclass = SNDRV_PCM_SUBCLASS_GENERIC_MIX; strcpy(pcm->name, "p16v"); emu->pcm_p16v = pcm; + emu->p16v_interrupt = snd_p16v_interrupt; for(substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; substream; diff --git a/sound/pci/emu10k1/p16v.h b/sound/pci/emu10k1/p16v.h index 3cdafa311617..9d429ad1feff 100644 --- a/sound/pci/emu10k1/p16v.h +++ b/sound/pci/emu10k1/p16v.h @@ -64,7 +64,7 @@ */ /********************************************************************************************************/ -/* Audigy2 P16V pointer-offset register set, accessed through the PTR2 and DATA2 registers */ +/* Audigy2 P16V pointer-offset register set, accessed through the PTR2 and DATA2 registers */ /********************************************************************************************************/ /* The sample rate of the SPDIF outputs is set by modifying a register in the EMU10K2 PTR register A_SPDIF_SAMPLERATE. diff --git a/sound/pci/emu10k1/p17v.h b/sound/pci/emu10k1/p17v.h index 3a6568346fad..d4ada1c430c8 100644 --- a/sound/pci/emu10k1/p17v.h +++ b/sound/pci/emu10k1/p17v.h @@ -6,8 +6,8 @@ */ /******************************************************************************/ -/* Audigy2Value Tina (P17V) pointer-offset register set, - * accessed through the PTR20 and DATA24 registers */ +/* Audigy2Value Tina (P17V) pointer-offset register set, */ +/* accessed through the PTR2 and DATA2 registers */ /******************************************************************************/ /* 00 - 07: Not used */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 81c4a45254ff..3226691ac923 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -227,6 +227,7 @@ enum { AZX_DRIVER_ATI, AZX_DRIVER_ATIHDMI, AZX_DRIVER_ATIHDMI_NS, + AZX_DRIVER_GFHDMI, AZX_DRIVER_VIA, AZX_DRIVER_SIS, AZX_DRIVER_ULI, @@ -328,14 +329,15 @@ enum { #define needs_eld_notify_link(chip) false #endif -#define CONTROLLER_IN_GPU(pci) (((pci)->device == 0x0a0c) || \ +#define CONTROLLER_IN_GPU(pci) (((pci)->vendor == 0x8086) && \ + (((pci)->device == 0x0a0c) || \ ((pci)->device == 0x0c0c) || \ ((pci)->device == 0x0d0c) || \ ((pci)->device == 0x160c) || \ ((pci)->device == 0x490d) || \ ((pci)->device == 0x4f90) || \ ((pci)->device == 0x4f91) || \ - ((pci)->device == 0x4f92)) + ((pci)->device == 0x4f92))) #define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98) @@ -348,6 +350,7 @@ static const char * const driver_short_names[] = { [AZX_DRIVER_ATI] = "HDA ATI SB", [AZX_DRIVER_ATIHDMI] = "HDA ATI HDMI", [AZX_DRIVER_ATIHDMI_NS] = "HDA ATI HDMI", + [AZX_DRIVER_GFHDMI] = "HDA GF HDMI", [AZX_DRIVER_VIA] = "HDA VIA VT82xx", [AZX_DRIVER_SIS] = "HDA SIS966", [AZX_DRIVER_ULI] = "HDA ULI M5461", @@ -1742,6 +1745,12 @@ static int default_bdl_pos_adj(struct azx *chip) } switch (chip->driver_type) { + /* + * increase the bdl size for Glenfly Gpus for hardware + * limitation on hdac interrupt interval + */ + case AZX_DRIVER_GFHDMI: + return 128; case AZX_DRIVER_ICH: case AZX_DRIVER_PCH: return 1; @@ -1857,6 +1866,12 @@ static int azx_first_init(struct azx *chip) pci_write_config_dword(pci, PCI_BASE_ADDRESS_1, 0); } #endif + /* + * Fix response write request not synced to memory when handle + * hdac interrupt on Glenfly Gpus + */ + if (chip->driver_type == AZX_DRIVER_GFHDMI) + bus->polling_mode = 1; err = pcim_iomap_regions(pci, 1 << 0, "ICH HD audio"); if (err < 0) @@ -1958,6 +1973,7 @@ static int azx_first_init(struct azx *chip) chip->playback_streams = ATIHDMI_NUM_PLAYBACK; chip->capture_streams = ATIHDMI_NUM_CAPTURE; break; + case AZX_DRIVER_GFHDMI: case AZX_DRIVER_GENERIC: default: chip->playback_streams = ICH6_NUM_PLAYBACK; @@ -2527,6 +2543,9 @@ static const struct pci_device_id azx_ids[] = { /* Meteorlake-P */ { PCI_DEVICE(0x8086, 0x7e28), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* Lunarlake-P */ + { PCI_DEVICE(0x8086, 0xa828), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Broxton-P(Apollolake) */ { PCI_DEVICE(0x8086, 0x5a98), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_BROXTON }, @@ -2723,6 +2742,12 @@ static const struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x1002, 0xab38), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS | AZX_DCAPS_PM_RUNTIME }, + /* GLENFLY */ + { PCI_DEVICE(0x6766, PCI_ANY_ID), + .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, + .class_mask = 0xffffff, + .driver_data = AZX_DRIVER_GFHDMI | AZX_DCAPS_POSFIX_LPIB | + AZX_DCAPS_NO_MSI | AZX_DCAPS_NO_64BIT }, /* VIA VT8251/VT8237A */ { PCI_DEVICE(0x1106, 0x3288), .driver_data = AZX_DRIVER_VIA }, /* VIA GFX VT7122/VX900 */ diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index acde4cd58785..099722ebaed8 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -4228,8 +4228,10 @@ static int tuning_ctl_set(struct hda_codec *codec, hda_nid_t nid, for (i = 0; i < TUNING_CTLS_COUNT; i++) if (nid == ca0132_tuning_ctls[i].nid) - break; + goto found; + return -EINVAL; +found: snd_hda_power_up(codec); dspio_set_param(codec, ca0132_tuning_ctls[i].mid, 0x20, ca0132_tuning_ctls[i].req, diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 75e1d00074b9..a889cccdd607 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -980,7 +980,10 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x3905, "Lenovo G50-30", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x390b, "Lenovo G50-80", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), - SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_PINCFG_LENOVO_NOTEBOOK), + /* NOTE: we'd need to extend the quirk for 17aa:3977 as the same + * PCI SSID is used on multiple Lenovo models + */ + SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo G50-70", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK_VENDOR(0x17aa, "Thinkpad", CXT_FIXUP_THINKPAD_ACPI), @@ -1003,6 +1006,7 @@ static const struct hda_model_fixup cxt5066_fixup_models[] = { { .id = CXT_FIXUP_MUTE_LED_GPIO, .name = "mute-led-gpio" }, { .id = CXT_FIXUP_HP_ZBOOK_MUTE_LED, .name = "hp-zbook-mute-led" }, { .id = CXT_FIXUP_HP_MIC_NO_PRESENCE, .name = "hp-mic-fix" }, + { .id = CXT_PINCFG_LENOVO_NOTEBOOK, .name = "lenovo-20149" }, {} }; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 9ea633fe9339..64a944016c78 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -81,6 +81,7 @@ struct hdmi_spec_per_pin { struct delayed_work work; struct hdmi_pcm *pcm; /* pointer to spec->pcm_rec[n] dynamically*/ int pcm_idx; /* which pcm is attached. -1 means no pcm is attached */ + int prev_pcm_idx; /* previously assigned pcm index */ int repoll_count; bool setup; /* the stream has been set up by prepare callback */ bool silent_stream; @@ -1380,9 +1381,17 @@ static void hdmi_attach_hda_pcm(struct hdmi_spec *spec, /* pcm already be attached to the pin */ if (per_pin->pcm) return; + /* try the previously used slot at first */ + idx = per_pin->prev_pcm_idx; + if (idx >= 0) { + if (!test_bit(idx, &spec->pcm_bitmap)) + goto found; + per_pin->prev_pcm_idx = -1; /* no longer valid, clear it */ + } idx = hdmi_find_pcm_slot(spec, per_pin); if (idx == -EBUSY) return; + found: per_pin->pcm_idx = idx; per_pin->pcm = get_hdmi_pcm(spec, idx); set_bit(idx, &spec->pcm_bitmap); @@ -1398,6 +1407,7 @@ static void hdmi_detach_hda_pcm(struct hdmi_spec *spec, return; idx = per_pin->pcm_idx; per_pin->pcm_idx = -1; + per_pin->prev_pcm_idx = idx; /* remember the previous index */ per_pin->pcm = NULL; if (idx >= 0 && idx < spec->pcm_used) clear_bit(idx, &spec->pcm_bitmap); @@ -1924,6 +1934,7 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) per_pin->pcm = NULL; per_pin->pcm_idx = -1; + per_pin->prev_pcm_idx = -1; per_pin->pin_nid = pin_nid; per_pin->pin_nid_idx = spec->num_nids; per_pin->dev_id = i; @@ -2093,10 +2104,6 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, goto unlock; } - if (snd_BUG_ON(pin_idx < 0)) { - err = -EINVAL; - goto unlock; - } per_pin = get_pin(spec, pin_idx); /* Verify pin:cvt selections to avoid silent audio after S3. @@ -2188,13 +2195,13 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, snd_hda_spdif_ctls_unassign(codec, pcm_idx); clear_bit(pcm_idx, &spec->pcm_in_use); pin_idx = hinfo_to_pin_index(codec, hinfo); + /* + * In such a case, return 0 to match the behavior in + * hdmi_pcm_open() + */ if (pin_idx < 0) goto unlock; - if (snd_BUG_ON(pin_idx < 0)) { - err = -EINVAL; - goto unlock; - } per_pin = get_pin(spec, pin_idx); if (spec->dyn_pin_out) { @@ -4478,6 +4485,22 @@ static int patch_via_hdmi(struct hda_codec *codec) return patch_simple_hdmi(codec, VIAHDMI_CVT_NID, VIAHDMI_PIN_NID); } +static int patch_gf_hdmi(struct hda_codec *codec) +{ + int err; + + err = patch_generic_hdmi(codec); + if (err) + return err; + + /* + * Glenfly GPUs have two codecs, stream switches from one codec to + * another, need to do actual clean-ups in codec_cleanup_stream + */ + codec->no_sticky_stream = 1; + return 0; +} + /* * patch entries */ @@ -4568,6 +4591,12 @@ HDA_CODEC_ENTRY(0x10de009f, "GPU 9f HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de00a0, "GPU a0 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de8001, "MCP73 HDMI", patch_nvhdmi_2ch), HDA_CODEC_ENTRY(0x10de8067, "MCP67/68 HDMI", patch_nvhdmi_2ch), +HDA_CODEC_ENTRY(0x67663d82, "Arise 82 HDMI/DP", patch_gf_hdmi), +HDA_CODEC_ENTRY(0x67663d83, "Arise 83 HDMI/DP", patch_gf_hdmi), +HDA_CODEC_ENTRY(0x67663d84, "Arise 84 HDMI/DP", patch_gf_hdmi), +HDA_CODEC_ENTRY(0x67663d85, "Arise 85 HDMI/DP", patch_gf_hdmi), +HDA_CODEC_ENTRY(0x67663d86, "Arise 86 HDMI/DP", patch_gf_hdmi), +HDA_CODEC_ENTRY(0x67663d87, "Arise 87 HDMI/DP", patch_gf_hdmi), HDA_CODEC_ENTRY(0x11069f80, "VX900 HDMI/DP", patch_via_hdmi), HDA_CODEC_ENTRY(0x11069f81, "VX900 HDMI/DP", patch_via_hdmi), HDA_CODEC_ENTRY(0x11069f84, "VX11 HDMI/DP", patch_generic_hdmi), @@ -4593,7 +4622,7 @@ HDA_CODEC_ENTRY(0x80862814, "DG1 HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862815, "Alderlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862816, "Rocketlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862818, "Raptorlake HDMI", patch_i915_tgl_hdmi), -HDA_CODEC_ENTRY(0x80862819, "DG2 HDMI", patch_i915_adlp_hdmi), +HDA_CODEC_ENTRY(0x80862819, "DG2 HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281c, "Alderlake-P HDMI", patch_i915_adlp_hdmi), diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3c629f4ae080..068ce0db9562 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2624,6 +2624,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1462, 0xda57, "MSI Z270-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX), + SND_PCI_QUIRK(0x1558, 0x3702, "Clevo X370SN[VW]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x50d3, "Clevo PC50[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x65d1, "Clevo PB51[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x65d2, "Clevo PB51R[CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), @@ -2631,6 +2632,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x65e5, "Clevo PC50D[PRS](?:-D|-G)?", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x65f1, "Clevo PC50HS", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x65f5, "Clevo PD50PN[NRT]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x66a2, "Clevo PE60RNE", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x67d1, "Clevo PB71[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x67e1, "Clevo PB71[DE][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x67e5, "Clevo PC70D[PRS](?:-D|-G)?", ALC1220_FIXUP_CLEVO_PB51ED_PINS), @@ -2651,6 +2653,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x96e1, "Clevo P960[ER][CDFN]-K", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1558, 0x97e1, "Clevo P970[ER][CDFN]", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1558, 0x97e2, "Clevo P970RC-M", ALC1220_FIXUP_CLEVO_P950), + SND_PCI_QUIRK(0x1558, 0xd502, "Clevo PD50SNE", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", ALC882_FIXUP_LENOVO_Y530), @@ -6957,6 +6960,8 @@ enum { ALC269_FIXUP_DELL_M101Z, ALC269_FIXUP_SKU_IGNORE, ALC269_FIXUP_ASUS_G73JW, + ALC269_FIXUP_ASUS_N7601ZM_PINS, + ALC269_FIXUP_ASUS_N7601ZM, ALC269_FIXUP_LENOVO_EAPD, ALC275_FIXUP_SONY_HWEQ, ALC275_FIXUP_SONY_DISABLE_AAMIX, @@ -7253,6 +7258,29 @@ static const struct hda_fixup alc269_fixups[] = { { } } }, + [ALC269_FIXUP_ASUS_N7601ZM_PINS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03A11050 }, + { 0x1a, 0x03A11C30 }, + { 0x21, 0x03211420 }, + { } + } + }, + [ALC269_FIXUP_ASUS_N7601ZM] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + {0x20, AC_VERB_SET_COEF_INDEX, 0x62}, + {0x20, AC_VERB_SET_PROC_COEF, 0xa007}, + {0x20, AC_VERB_SET_COEF_INDEX, 0x10}, + {0x20, AC_VERB_SET_PROC_COEF, 0x8420}, + {0x20, AC_VERB_SET_COEF_INDEX, 0x0f}, + {0x20, AC_VERB_SET_PROC_COEF, 0x7774}, + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_ASUS_N7601ZM_PINS, + }, [ALC269_FIXUP_LENOVO_EAPD] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -9260,7 +9288,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0a62, "Dell Precision 5560", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x0a9d, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0a9e, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x0ac9, "Dell Precision 3260", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x1028, 0x0b19, "Dell XPS 15 9520", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x0b1a, "Dell Precision 5570", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x0b37, "Dell Inspiron 16 Plus 7620 2-in-1", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS), @@ -9401,6 +9428,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8898, "HP EliteBook 845 G8 Notebook PC", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x103c, 0x88d0, "HP Pavilion 15-eh1xxx (mainboard 88D0)", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8902, "HP OMEN 16", ALC285_FIXUP_HP_MUTE_LED), + SND_PCI_QUIRK(0x103c, 0x8919, "HP Pavilion Aero Laptop 13-be0xxx", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x896d, "HP ZBook Firefly 16 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x896e, "HP EliteBook x360 830 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8971, "HP EliteBook 830 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), @@ -9441,12 +9469,15 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8b47, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b5d, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x8b5e, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), + SND_PCI_QUIRK(0x103c, 0x8b65, "HP ProBook 455 15.6 inch G10 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), + SND_PCI_QUIRK(0x103c, 0x8b66, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x8b7a, "HP", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b7d, "HP", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b87, "HP", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b8a, "HP", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b8b, "HP", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b8d, "HP", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8b8f, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b92, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8bf0, "HP", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), @@ -9461,6 +9492,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1271, "ASUS X430UN", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1290, "ASUS X441SA", ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x12a0, "ASUS X441UV", ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1043, 0x12a3, "Asus N7691ZM", ALC269_FIXUP_ASUS_N7601ZM), SND_PCI_QUIRK(0x1043, 0x12af, "ASUS UX582ZS", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x12e0, "ASUS X541SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x12f0, "ASUS X541UV", ALC256_FIXUP_ASUS_MIC), @@ -9469,6 +9501,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), SND_PCI_QUIRK(0x1043, 0x1662, "ASUS GV301QH", ALC294_FIXUP_ASUS_DUAL_SPK), + SND_PCI_QUIRK(0x1043, 0x1683, "ASUS UM3402YAR", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x16b2, "ASUS GU603", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x1740, "ASUS UX430UA", ALC295_FIXUP_ASUS_DACS), @@ -9539,6 +9572,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc830, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_AMP), SND_PCI_QUIRK(0x144d, 0xc832, "Samsung Galaxy Book Flex Alpha (NP730QCJ)", ALC256_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xca03, "Samsung Galaxy Book2 Pro 360 (NP930QED)", ALC298_FIXUP_SAMSUNG_AMP), + SND_PCI_QUIRK(0x144d, 0xc868, "Samsung Galaxy Book2 Pro (NP930XED)", ALC298_FIXUP_SAMSUNG_AMP), SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), @@ -9573,6 +9607,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x5101, "Clevo S510WU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x5157, "Clevo W517GU1", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x51a1, "Clevo NS50MU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x5630, "Clevo NP50RNJS", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x70a1, "Clevo NB70T[HJK]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x70b3, "Clevo NK70SB", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x70f2, "Clevo NH79EPY", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), @@ -9607,6 +9642,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x971d, "Clevo N970T[CDF]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xa500, "Clevo NL5[03]RU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xa600, "Clevo NL50NU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0xa671, "Clevo NP70SN[CDE]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xb018, "Clevo NP50D[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xb019, "Clevo NH77D[BE]Q", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xb022, "Clevo NH77D[DC][QW]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), @@ -9655,6 +9691,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x22f1, "Thinkpad", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x22f2, "Thinkpad", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x22f3, "Thinkpad", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x2316, "Thinkpad P1 Gen 6", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x2317, "Thinkpad P1 Gen 6", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x2318, "Thinkpad Z13 Gen2", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x2319, "Thinkpad Z16 Gen2", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x231a, "Thinkpad Z16 Gen2", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x310c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), @@ -9707,6 +9748,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x511e, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x511f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), + SND_PCI_QUIRK(0x17aa, 0x9e56, "Lenovo ZhaoYang CF4620Z", ALC286_FIXUP_SONY_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1849, 0x1233, "ASRock NUC Box 1100", ALC233_FIXUP_NO_AUDIO_JACK), SND_PCI_QUIRK(0x1849, 0xa233, "Positivo Master C6300", ALC269_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MACH-WX9", ALC256_FIXUP_HUAWEI_MACH_WX9_PINS), diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a794a01a68ca..61258b0aac8d 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1707,6 +1707,7 @@ static const struct snd_pci_quirk stac925x_fixup_tbl[] = { }; static const struct hda_pintbl ref92hd73xx_pin_configs[] = { + // Port A-H { 0x0a, 0x02214030 }, { 0x0b, 0x02a19040 }, { 0x0c, 0x01a19020 }, @@ -1715,9 +1716,12 @@ static const struct hda_pintbl ref92hd73xx_pin_configs[] = { { 0x0f, 0x01014010 }, { 0x10, 0x01014020 }, { 0x11, 0x01014030 }, + // CD in { 0x12, 0x02319040 }, + // Digial Mic ins { 0x13, 0x90a000f0 }, { 0x14, 0x90a000f0 }, + // Digital outs { 0x22, 0x01452050 }, { 0x23, 0x01452050 }, {} @@ -1758,6 +1762,7 @@ static const struct hda_pintbl alienware_m17x_pin_configs[] = { }; static const struct hda_pintbl intel_dg45id_pin_configs[] = { + // Analog outputs { 0x0a, 0x02214230 }, { 0x0b, 0x02A19240 }, { 0x0c, 0x01013214 }, @@ -1765,6 +1770,9 @@ static const struct hda_pintbl intel_dg45id_pin_configs[] = { { 0x0e, 0x01A19250 }, { 0x0f, 0x01011212 }, { 0x10, 0x01016211 }, + // Digital output + { 0x22, 0x01451380 }, + { 0x23, 0x40f000f0 }, {} }; @@ -1955,6 +1963,8 @@ static const struct snd_pci_quirk stac92hd73xx_fixup_tbl[] = { "DFI LanParty", STAC_92HD73XX_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, "DFI LanParty", STAC_92HD73XX_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x5001, + "Intel DP45SG", STAC_92HD73XX_INTEL), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x5002, "Intel DG45ID", STAC_92HD73XX_INTEL), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x5003, diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index fa1812e7a49d..267c7848974a 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6146,12 +6146,6 @@ static int snd_hdspm_hwdep_dummy_op(struct snd_hwdep *hw, struct file *file) return 0; } -static inline int copy_u32_le(void __user *dest, void __iomem *src) -{ - u32 val = readl(src); - return copy_to_user(dest, &val, 4); -} - static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigned int cmd, unsigned long arg) { diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c index 1e198e4d57b8..b033bd290940 100644 --- a/sound/pci/ymfpci/ymfpci.c +++ b/sound/pci/ymfpci/ymfpci.c @@ -98,8 +98,10 @@ static int snd_ymfpci_create_gameport(struct snd_ymfpci *chip, int dev, case 0x204: legacy_ctrl2 |= 2 << 6; break; case 0x205: legacy_ctrl2 |= 3 << 6; break; default: - dev_err(chip->card->dev, - "invalid joystick port %#x", io_port); + if (io_port > 0) + dev_err(chip->card->dev, + "The %s does not support arbitrary IO ports for the game port (requested 0x%x)\n", + chip->card->shortname, (unsigned int)io_port); return -EINVAL; } } @@ -170,7 +172,7 @@ static int snd_card_ymfpci_probe(struct pci_dev *pci, return -ENOENT; } - err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, + err = snd_devm_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, sizeof(*chip), &card); if (err < 0) return err; @@ -186,6 +188,13 @@ static int snd_card_ymfpci_probe(struct pci_dev *pci, default: model = str = "???"; break; } + strcpy(card->driver, str); + sprintf(card->shortname, "Yamaha %s (%s)", model, str); + sprintf(card->longname, "%s at 0x%lx, irq %i", + card->shortname, + chip->reg_area_phys, + chip->irq); + legacy_ctrl = 0; legacy_ctrl2 = 0x0800; /* SBEN = 0, SMOD = 01, LAD = 0 */ @@ -218,7 +227,13 @@ static int snd_card_ymfpci_probe(struct pci_dev *pci, case 0x398: legacy_ctrl2 |= 1; break; case 0x3a0: legacy_ctrl2 |= 2; break; case 0x3a8: legacy_ctrl2 |= 3; break; - default: fm_port[dev] = 0; break; + default: + if (fm_port[dev] > 0) + dev_err(card->dev, + "The %s does not support arbitrary IO ports for FM (requested 0x%x)\n", + card->shortname, (unsigned int)fm_port[dev]); + fm_port[dev] = 0; + break; } if (fm_port[dev] > 0) fm_res = devm_request_region(&pci->dev, fm_port[dev], @@ -234,7 +249,13 @@ static int snd_card_ymfpci_probe(struct pci_dev *pci, case 0x300: legacy_ctrl2 |= 1 << 4; break; case 0x332: legacy_ctrl2 |= 2 << 4; break; case 0x334: legacy_ctrl2 |= 3 << 4; break; - default: mpu_port[dev] = 0; break; + default: + if (mpu_port[dev] > 0) + dev_err(card->dev, + "The %s does not support arbitrary IO ports for MPU-401 (requested 0x%x)\n", + card->shortname, (unsigned int)mpu_port[dev]); + mpu_port[dev] = 0; + break; } if (mpu_port[dev] > 0) mpu_res = devm_request_region(&pci->dev, mpu_port[dev], @@ -257,12 +278,6 @@ static int snd_card_ymfpci_probe(struct pci_dev *pci, if (err < 0) return err; - strcpy(card->driver, str); - sprintf(card->shortname, "Yamaha %s (%s)", model, str); - sprintf(card->longname, "%s at 0x%lx, irq %i", - card->shortname, - chip->reg_area_phys, - chip->irq); err = snd_ymfpci_pcm(chip, 0); if (err < 0) return err; @@ -337,11 +352,9 @@ static struct pci_driver ymfpci_driver = { .name = KBUILD_MODNAME, .id_table = snd_ymfpci_ids, .probe = snd_card_ymfpci_probe, -#ifdef CONFIG_PM_SLEEP .driver = { - .pm = &snd_ymfpci_pm, + .pm = pm_sleep_ptr(&snd_ymfpci_pm), }, -#endif }; module_pci_driver(ymfpci_driver); diff --git a/sound/pci/ymfpci/ymfpci.h b/sound/pci/ymfpci/ymfpci.h index 66968776478a..a408785cfa1b 100644 --- a/sound/pci/ymfpci/ymfpci.h +++ b/sound/pci/ymfpci/ymfpci.h @@ -268,6 +268,49 @@ struct snd_ymfpci_pcm { u32 shift; }; +static const int saved_regs_index[] = { + /* spdif */ + YDSXGR_SPDIFOUTCTRL, + YDSXGR_SPDIFOUTSTATUS, + YDSXGR_SPDIFINCTRL, + /* volumes */ + YDSXGR_PRIADCLOOPVOL, + YDSXGR_NATIVEDACINVOL, + YDSXGR_NATIVEDACOUTVOL, + YDSXGR_BUF441OUTVOL, + YDSXGR_NATIVEADCINVOL, + YDSXGR_SPDIFLOOPVOL, + YDSXGR_SPDIFOUTVOL, + YDSXGR_ZVOUTVOL, + YDSXGR_LEGACYOUTVOL, + /* address bases */ + YDSXGR_PLAYCTRLBASE, + YDSXGR_RECCTRLBASE, + YDSXGR_EFFCTRLBASE, + YDSXGR_WORKBASE, + /* capture set up */ + YDSXGR_MAPOFREC, + YDSXGR_RECFORMAT, + YDSXGR_RECSLOTSR, + YDSXGR_ADCFORMAT, + YDSXGR_ADCSLOTSR, +}; +#define YDSXGR_NUM_SAVED_REGS ARRAY_SIZE(saved_regs_index) + +static const int pci_saved_regs_index[] = { + /* All Chips */ + PCIR_DSXG_LEGACY, + PCIR_DSXG_ELEGACY, + /* YMF 744/754 */ + PCIR_DSXG_FMBASE, + PCIR_DSXG_SBBASE, + PCIR_DSXG_MPU401BASE, + PCIR_DSXG_JOYBASE, +}; +#define DSXG_PCI_NUM_SAVED_REGS ARRAY_SIZE(pci_saved_regs_index) +#define DSXG_PCI_NUM_SAVED_LEGACY_REGS 2 +static_assert(DSXG_PCI_NUM_SAVED_LEGACY_REGS <= DSXG_PCI_NUM_SAVED_REGS); + struct snd_ymfpci { int irq; @@ -276,7 +319,7 @@ struct snd_ymfpci { unsigned long reg_area_phys; void __iomem *reg_area_virt; - unsigned short old_legacy_ctrl; + u16 old_legacy_ctrl; #ifdef SUPPORT_JOYSTICK struct gameport *gameport; #endif @@ -345,17 +388,14 @@ struct snd_ymfpci { const struct firmware *dsp_microcode; const struct firmware *controller_microcode; -#ifdef CONFIG_PM_SLEEP - u32 *saved_regs; + u32 saved_regs[YDSXGR_NUM_SAVED_REGS]; u32 saved_ydsxgr_mode; - u16 saved_dsxg_legacy; - u16 saved_dsxg_elegacy; -#endif + u16 saved_dsxg_pci_regs[DSXG_PCI_NUM_SAVED_REGS]; }; int snd_ymfpci_create(struct snd_card *card, struct pci_dev *pci, - unsigned short old_legacy_ctrl); + u16 old_legacy_ctrl); void snd_ymfpci_free_gameport(struct snd_ymfpci *chip); extern const struct dev_pm_ops snd_ymfpci_pm; diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index c80114c0ad7b..6971eec45a4d 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -31,11 +31,6 @@ static void snd_ymfpci_irq_wait(struct snd_ymfpci *chip); -static inline u8 snd_ymfpci_readb(struct snd_ymfpci *chip, u32 offset) -{ - return readb(chip->reg_area_virt + offset); -} - static inline void snd_ymfpci_writeb(struct snd_ymfpci *chip, u32 offset, u8 val) { writeb(val, chip->reg_area_virt + offset); @@ -2165,7 +2160,7 @@ static int snd_ymfpci_memalloc(struct snd_ymfpci *chip) chip->work_base = ptr; chip->work_base_addr = ptr_addr; - snd_BUG_ON(ptr + chip->work_size != + snd_BUG_ON(ptr + PAGE_ALIGN(chip->work_size) != chip->work_ptr->area + chip->work_ptr->bytes); snd_ymfpci_writel(chip, YDSXGR_PLAYCTRLBASE, chip->bank_base_playback_addr); @@ -2219,57 +2214,33 @@ static void snd_ymfpci_free(struct snd_card *card) snd_ymfpci_free_gameport(chip); - pci_write_config_word(chip->pci, 0x40, chip->old_legacy_ctrl); + pci_write_config_word(chip->pci, PCIR_DSXG_LEGACY, chip->old_legacy_ctrl); release_firmware(chip->dsp_microcode); release_firmware(chip->controller_microcode); } -#ifdef CONFIG_PM_SLEEP -static const int saved_regs_index[] = { - /* spdif */ - YDSXGR_SPDIFOUTCTRL, - YDSXGR_SPDIFOUTSTATUS, - YDSXGR_SPDIFINCTRL, - /* volumes */ - YDSXGR_PRIADCLOOPVOL, - YDSXGR_NATIVEDACINVOL, - YDSXGR_NATIVEDACOUTVOL, - YDSXGR_BUF441OUTVOL, - YDSXGR_NATIVEADCINVOL, - YDSXGR_SPDIFLOOPVOL, - YDSXGR_SPDIFOUTVOL, - YDSXGR_ZVOUTVOL, - YDSXGR_LEGACYOUTVOL, - /* address bases */ - YDSXGR_PLAYCTRLBASE, - YDSXGR_RECCTRLBASE, - YDSXGR_EFFCTRLBASE, - YDSXGR_WORKBASE, - /* capture set up */ - YDSXGR_MAPOFREC, - YDSXGR_RECFORMAT, - YDSXGR_RECSLOTSR, - YDSXGR_ADCFORMAT, - YDSXGR_ADCSLOTSR, -}; -#define YDSXGR_NUM_SAVED_REGS ARRAY_SIZE(saved_regs_index) - static int snd_ymfpci_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); struct snd_ymfpci *chip = card->private_data; - unsigned int i; - + unsigned int i, legacy_reg_count = DSXG_PCI_NUM_SAVED_LEGACY_REGS; + + if (chip->pci->device >= 0x0010) /* YMF 744/754 */ + legacy_reg_count = DSXG_PCI_NUM_SAVED_REGS; + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); snd_ac97_suspend(chip->ac97); + for (i = 0; i < YDSXGR_NUM_SAVED_REGS; i++) chip->saved_regs[i] = snd_ymfpci_readl(chip, saved_regs_index[i]); + chip->saved_ydsxgr_mode = snd_ymfpci_readl(chip, YDSXGR_MODE); - pci_read_config_word(chip->pci, PCIR_DSXG_LEGACY, - &chip->saved_dsxg_legacy); - pci_read_config_word(chip->pci, PCIR_DSXG_ELEGACY, - &chip->saved_dsxg_elegacy); + + for (i = 0; i < legacy_reg_count; i++) + pci_read_config_word(chip->pci, pci_saved_regs_index[i], + chip->saved_dsxg_pci_regs + i); + snd_ymfpci_writel(chip, YDSXGR_NATIVEDACOUTVOL, 0); snd_ymfpci_writel(chip, YDSXGR_BUF441OUTVOL, 0); snd_ymfpci_disable_dsp(chip); @@ -2281,7 +2252,10 @@ static int snd_ymfpci_resume(struct device *dev) struct pci_dev *pci = to_pci_dev(dev); struct snd_card *card = dev_get_drvdata(dev); struct snd_ymfpci *chip = card->private_data; - unsigned int i; + unsigned int i, legacy_reg_count = DSXG_PCI_NUM_SAVED_LEGACY_REGS; + + if (chip->pci->device >= 0x0010) /* YMF 744/754 */ + legacy_reg_count = DSXG_PCI_NUM_SAVED_REGS; snd_ymfpci_aclink_reset(pci); snd_ymfpci_codec_ready(chip, 0); @@ -2293,10 +2267,9 @@ static int snd_ymfpci_resume(struct device *dev) snd_ac97_resume(chip->ac97); - pci_write_config_word(chip->pci, PCIR_DSXG_LEGACY, - chip->saved_dsxg_legacy); - pci_write_config_word(chip->pci, PCIR_DSXG_ELEGACY, - chip->saved_dsxg_elegacy); + for (i = 0; i < legacy_reg_count; i++) + pci_write_config_word(chip->pci, pci_saved_regs_index[i], + chip->saved_dsxg_pci_regs[i]); /* start hw again */ if (chip->start_count > 0) { @@ -2309,12 +2282,11 @@ static int snd_ymfpci_resume(struct device *dev) return 0; } -SIMPLE_DEV_PM_OPS(snd_ymfpci_pm, snd_ymfpci_suspend, snd_ymfpci_resume); -#endif /* CONFIG_PM_SLEEP */ +DEFINE_SIMPLE_DEV_PM_OPS(snd_ymfpci_pm, snd_ymfpci_suspend, snd_ymfpci_resume); int snd_ymfpci_create(struct snd_card *card, struct pci_dev *pci, - unsigned short old_legacy_ctrl) + u16 old_legacy_ctrl) { struct snd_ymfpci *chip = card->private_data; int err; @@ -2379,13 +2351,6 @@ int snd_ymfpci_create(struct snd_card *card, if (err < 0) return err; -#ifdef CONFIG_PM_SLEEP - chip->saved_regs = devm_kmalloc_array(&pci->dev, YDSXGR_NUM_SAVED_REGS, - sizeof(u32), GFP_KERNEL); - if (!chip->saved_regs) - return -ENOMEM; -#endif - snd_ymfpci_proc_init(card, chip); return 0; diff --git a/sound/ppc/tumbler.c b/sound/ppc/tumbler.c index f3f8ad7c3df8..12f1e10db1c4 100644 --- a/sound/ppc/tumbler.c +++ b/sound/ppc/tumbler.c @@ -1361,9 +1361,9 @@ int snd_pmac_tumbler_init(struct snd_pmac *chip) for_each_child_of_node(chip->node, np) { if (of_node_name_eq(np, "sound")) { - if (of_get_property(np, "has-anded-reset", NULL)) + if (of_property_read_bool(np, "has-anded-reset")) mix->anded_reset = 1; - if (of_get_property(np, "layout-id", NULL)) + if (of_property_present(np, "layout-id")) mix->reset_on_sleep = 0; of_node_put(np); break; diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index f90a6a7ba83b..fde055c6c894 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -31,7 +31,7 @@ static int max98373_dac_event(struct snd_soc_dapm_widget *w, MAX98373_GLOBAL_EN_MASK, 1); usleep_range(30000, 31000); break; - case SND_SOC_DAPM_POST_PMD: + case SND_SOC_DAPM_PRE_PMD: regmap_update_bits(max98373->regmap, MAX98373_R20FF_GLOBAL_SHDN, MAX98373_GLOBAL_EN_MASK, 0); @@ -64,7 +64,7 @@ static const struct snd_kcontrol_new max98373_spkfb_control = static const struct snd_soc_dapm_widget max98373_dapm_widgets[] = { SND_SOC_DAPM_DAC_E("Amp Enable", "HiFi Playback", MAX98373_R202B_PCM_RX_EN, 0, 0, max98373_dac_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_MUX("DAI Sel Mux", SND_SOC_NOPM, 0, 0, &max98373_dai_controls), SND_SOC_DAPM_OUTPUT("BE_OUT"), diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index 3b81a465814a..05a7d1588d20 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -209,14 +209,19 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, be_chan = soc_component_to_pcm(component_be)->chan[substream->stream]; tmp_chan = be_chan; } - if (!tmp_chan) - tmp_chan = dma_request_slave_channel(dev_be, tx ? "tx" : "rx"); + if (!tmp_chan) { + tmp_chan = dma_request_chan(dev_be, tx ? "tx" : "rx"); + if (IS_ERR(tmp_chan)) { + dev_err(dev, "failed to request DMA channel for Back-End\n"); + return -EINVAL; + } + } /* * An EDMA DEV_TO_DEV channel is fixed and bound with DMA event of each * peripheral, unlike SDMA channel that is allocated dynamically. So no * need to configure dma_request and dma_request2, but get dma_chan of - * Back-End device directly via dma_request_slave_channel. + * Back-End device directly via dma_request_chan. */ if (!asrc->use_edma) { /* Get DMA request of Back-End */ diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 07d13dca852e..abdaffb00fbd 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1544,7 +1544,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx8qm_data = { .use_imx_pcm = true, .use_edma = true, .fifo_depth = 64, - .pins = 1, + .pins = 4, .reg_offset = 0, .mclk0_is_mclk1 = false, .flags = 0, diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index a010e7c38817..059eebf0a687 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -2349,10 +2349,12 @@ static int sof_ipc4_route_setup(struct snd_sof_dev *sdev, struct snd_sof_route * } if (!src_fw_module || !sink_fw_module) { - /* The NULL module will print as "(efault)" */ - dev_err(sdev->dev, "source %s or sink %s widget weren't set up properly\n", - src_fw_module->man4_module_entry.name, - sink_fw_module->man4_module_entry.name); + dev_err(sdev->dev, + "cannot bind %s -> %s, no firmware module for: %s%s\n", + src_widget->widget->name, sink_widget->widget->name, + src_fw_module ? "" : " source", + sink_fw_module ? "" : " sink"); + return -ENODEV; } diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c index c74ce8d414e7..2fdbc53ca715 100644 --- a/sound/soc/sof/pm.c +++ b/sound/soc/sof/pm.c @@ -188,6 +188,7 @@ static int sof_suspend(struct device *dev, bool runtime_suspend) const struct sof_ipc_tplg_ops *tplg_ops = sof_ipc_get_ops(sdev, tplg); pm_message_t pm_state; u32 target_state = snd_sof_dsp_power_target(sdev); + u32 old_state = sdev->dsp_power_state.state; int ret; /* do nothing if dsp suspend callback is not set */ @@ -197,7 +198,12 @@ static int sof_suspend(struct device *dev, bool runtime_suspend) if (runtime_suspend && !sof_ops(sdev)->runtime_suspend) return 0; - if (tplg_ops && tplg_ops->tear_down_all_pipelines) + /* we need to tear down pipelines only if the DSP hardware is + * active, which happens for PCI devices. if the device is + * suspended, it is brought back to full power and then + * suspended again + */ + if (tplg_ops && tplg_ops->tear_down_all_pipelines && (old_state == SOF_DSP_PM_D0)) tplg_ops->tear_down_all_pipelines(sdev, false); if (sdev->fw_state != SOF_FW_BOOT_COMPLETE) diff --git a/sound/usb/caiaq/input.c b/sound/usb/caiaq/input.c index 1e2cf2f08eec..84f26dce7f5d 100644 --- a/sound/usb/caiaq/input.c +++ b/sound/usb/caiaq/input.c @@ -804,6 +804,7 @@ int snd_usb_caiaq_input_init(struct snd_usb_caiaqdev *cdev) default: /* no input methods supported on this device */ + ret = -EINVAL; goto exit_free_idev; } diff --git a/sound/usb/card.c b/sound/usb/card.c index 26268ffb8274..f6e99ced8068 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -609,7 +609,6 @@ static int snd_usb_audio_create(struct usb_interface *intf, case USB_SPEED_LOW: case USB_SPEED_FULL: case USB_SPEED_HIGH: - case USB_SPEED_WIRELESS: case USB_SPEED_SUPER: case USB_SPEED_SUPER_PLUS: break; diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 419302e2057e..a385e85c4650 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -455,8 +455,8 @@ static void push_back_to_ready_list(struct snd_usb_endpoint *ep, * This function is used both for implicit feedback endpoints and in low- * latency playback mode. */ -void snd_usb_queue_pending_output_urbs(struct snd_usb_endpoint *ep, - bool in_stream_lock) +int snd_usb_queue_pending_output_urbs(struct snd_usb_endpoint *ep, + bool in_stream_lock) { bool implicit_fb = snd_usb_endpoint_implicit_feedback_sink(ep); @@ -480,7 +480,7 @@ void snd_usb_queue_pending_output_urbs(struct snd_usb_endpoint *ep, spin_unlock_irqrestore(&ep->lock, flags); if (ctx == NULL) - return; + break; /* copy over the length information */ if (implicit_fb) { @@ -495,11 +495,14 @@ void snd_usb_queue_pending_output_urbs(struct snd_usb_endpoint *ep, break; if (err < 0) { /* push back to ready list again for -EAGAIN */ - if (err == -EAGAIN) + if (err == -EAGAIN) { push_back_to_ready_list(ep, ctx); - else + break; + } + + if (!in_stream_lock) notify_xrun(ep); - return; + return -EPIPE; } err = usb_submit_urb(ctx->urb, GFP_ATOMIC); @@ -507,13 +510,16 @@ void snd_usb_queue_pending_output_urbs(struct snd_usb_endpoint *ep, usb_audio_err(ep->chip, "Unable to submit urb #%d: %d at %s\n", ctx->index, err, __func__); - notify_xrun(ep); - return; + if (!in_stream_lock) + notify_xrun(ep); + return -EPIPE; } set_bit(ctx->index, &ep->active_mask); atomic_inc(&ep->submitted_urbs); } + + return 0; } /* @@ -910,8 +916,9 @@ static int endpoint_set_interface(struct snd_usb_audio *chip, ep->iface, altset, ep->ep_num); err = usb_set_interface(chip->dev, ep->iface, altset); if (err < 0) { - usb_audio_err(chip, "%d:%d: usb_set_interface failed (%d)\n", - ep->iface, altset, err); + usb_audio_err_ratelimited( + chip, "%d:%d: usb_set_interface failed (%d)\n", + ep->iface, altset, err); return err; } @@ -1173,22 +1180,8 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep) */ if (usb_pipein(ep->pipe) || ep->implicit_fb_sync) { - urb_packs = packs_per_ms; - /* - * Wireless devices can poll at a max rate of once per 4ms. - * For dataintervals less than 5, increase the packet count to - * allow the host controller to use bursting to fill in the - * gaps. - */ - if (snd_usb_get_speed(chip->dev) == USB_SPEED_WIRELESS) { - int interval = ep->datainterval; - while (interval < 5) { - urb_packs <<= 1; - ++interval; - } - } /* make capture URBs <= 1 ms and smaller than a period */ - urb_packs = min(max_packs_per_urb, urb_packs); + urb_packs = min(max_packs_per_urb, packs_per_ms); while (urb_packs > 1 && urb_packs * maxsize >= ep->cur_period_bytes) urb_packs >>= 1; ep->nurbs = MAX_URBS; diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index 924f4351588c..c09f68ce08b1 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -52,7 +52,7 @@ int snd_usb_endpoint_implicit_feedback_sink(struct snd_usb_endpoint *ep); int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep, struct snd_urb_ctx *ctx, int idx, unsigned int avail); -void snd_usb_queue_pending_output_urbs(struct snd_usb_endpoint *ep, - bool in_stream_lock); +int snd_usb_queue_pending_output_urbs(struct snd_usb_endpoint *ep, + bool in_stream_lock); #endif /* __USBAUDIO_ENDPOINT_H */ diff --git a/sound/usb/format.c b/sound/usb/format.c index 405dc0bf6678..4b1c5ba121f3 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -39,8 +39,12 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, case UAC_VERSION_1: default: { struct uac_format_type_i_discrete_descriptor *fmt = _fmt; - if (format >= 64) - return 0; /* invalid format */ + if (format >= 64) { + usb_audio_info(chip, + "%u:%d: invalid format type 0x%llx is detected, processed as PCM\n", + fp->iface, fp->altsetting, format); + format = UAC_FORMAT_TYPE_I_PCM; + } sample_width = fmt->bBitResolution; sample_bytes = fmt->bSubframeSize; format = 1ULL << format; diff --git a/sound/usb/helper.c b/sound/usb/helper.c index a4410267bf70..bf80e55d013a 100644 --- a/sound/usb/helper.c +++ b/sound/usb/helper.c @@ -108,7 +108,6 @@ unsigned char snd_usb_parse_datainterval(struct snd_usb_audio *chip, { switch (snd_usb_get_speed(chip->dev)) { case USB_SPEED_HIGH: - case USB_SPEED_WIRELESS: case USB_SPEED_SUPER: case USB_SPEED_SUPER_PLUS: if (get_endpoint(alts, 0)->bInterval >= 1 && diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index d959da7a1afb..eec5232f9fb2 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1639,7 +1639,7 @@ static int snd_usb_pcm_playback_ack(struct snd_pcm_substream *substream) * outputs here */ if (!ep->active_mask) - snd_usb_queue_pending_output_urbs(ep, true); + return snd_usb_queue_pending_output_urbs(ep, true); return 0; } diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 271884e35003..efb4a3311cc5 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3884,6 +3884,64 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +{ + /* + * PIONEER DJ DDJ-800 + * PCM is 6 channels out, 6 channels in @ 44.1 fixed + * The Feedback for the output is the input + */ + USB_DEVICE_VENDOR_SPEC(0x2b73, 0x0029), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 6, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_44100, + .rate_min = 44100, + .rate_max = 44100, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 44100 } + } + }, + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 6, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x82, + .ep_idx = 1, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC| + USB_ENDPOINT_USAGE_IMPLICIT_FB, + .rates = SNDRV_PCM_RATE_44100, + .rate_min = 44100, + .rate_max = 44100, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 44100 } + } + }, + { + .ifnum = -1 + } + } + } +}, + /* * MacroSilicon MS2100/MS2106 based AV capture cards * diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index f5a8dca66457..38a85b2c9a73 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -67,6 +67,8 @@ struct snd_usb_audio { #define usb_audio_err(chip, fmt, args...) \ dev_err(&(chip)->dev->dev, fmt, ##args) +#define usb_audio_err_ratelimited(chip, fmt, args...) \ + dev_err_ratelimited(&(chip)->dev->dev, fmt, ##args) #define usb_audio_warn(chip, fmt, args...) \ dev_warn(&(chip)->dev->dev, fmt, ##args) #define usb_audio_info(chip, fmt, args...) \ diff --git a/tools/testing/selftests/alsa/mixer-test.c b/tools/testing/selftests/alsa/mixer-test.c index 05f1749ae19d..c95d63e553f4 100644 --- a/tools/testing/selftests/alsa/mixer-test.c +++ b/tools/testing/selftests/alsa/mixer-test.c @@ -63,6 +63,7 @@ static void find_controls(void) struct card_data *card_data; struct ctl_data *ctl_data; snd_config_t *config; + char *card_name, *card_longname; card = -1; if (snd_card_next(&card) < 0 || card < 0) @@ -84,6 +85,15 @@ static void find_controls(void) goto next_card; } + err = snd_card_get_name(card, &card_name); + if (err != 0) + card_name = "Unknown"; + err = snd_card_get_longname(card, &card_longname); + if (err != 0) + card_longname = "Unknown"; + ksft_print_msg("Card %d - %s (%s)\n", card, + card_name, card_longname); + /* Count controls */ snd_ctl_elem_list_malloc(&card_data->ctls); snd_ctl_elem_list(card_data->handle, card_data->ctls); @@ -422,6 +432,9 @@ static void test_ctl_name(struct ctl_data *ctl) bool name_ok = true; bool check; + ksft_print_msg("%d.%d %s\n", ctl->card->card, ctl->elem, + ctl->name); + /* Only boolean controls should end in Switch */ if (strend(ctl->name, " Switch")) { if (snd_ctl_elem_info_get_type(ctl->info) != SND_CTL_ELEM_TYPE_BOOLEAN) { @@ -445,6 +458,48 @@ static void test_ctl_name(struct ctl_data *ctl) ctl->card->card, ctl->elem); } +static void show_values(struct ctl_data *ctl, snd_ctl_elem_value_t *orig_val, + snd_ctl_elem_value_t *read_val) +{ + long long orig_int, read_int; + int i; + + for (i = 0; i < snd_ctl_elem_info_get_count(ctl->info); i++) { + switch (snd_ctl_elem_info_get_type(ctl->info)) { + case SND_CTL_ELEM_TYPE_BOOLEAN: + orig_int = snd_ctl_elem_value_get_boolean(orig_val, i); + read_int = snd_ctl_elem_value_get_boolean(read_val, i); + break; + + case SND_CTL_ELEM_TYPE_INTEGER: + orig_int = snd_ctl_elem_value_get_integer(orig_val, i); + read_int = snd_ctl_elem_value_get_integer(read_val, i); + break; + + case SND_CTL_ELEM_TYPE_INTEGER64: + orig_int = snd_ctl_elem_value_get_integer64(orig_val, + i); + read_int = snd_ctl_elem_value_get_integer64(read_val, + i); + break; + + case SND_CTL_ELEM_TYPE_ENUMERATED: + orig_int = snd_ctl_elem_value_get_enumerated(orig_val, + i); + read_int = snd_ctl_elem_value_get_enumerated(read_val, + i); + break; + + default: + return; + } + + ksft_print_msg("%s.%d orig %lld read %lld, is_volatile %d\n", + ctl->name, i, orig_int, read_int, + snd_ctl_elem_info_is_volatile(ctl->info)); + } +} + static bool show_mismatch(struct ctl_data *ctl, int index, snd_ctl_elem_value_t *read_val, snd_ctl_elem_value_t *expected_val) @@ -584,12 +639,14 @@ static int write_and_verify(struct ctl_data *ctl, if (err < 1) { ksft_print_msg("No event generated for %s\n", ctl->name); + show_values(ctl, initial_val, read_val); ctl->event_missing++; } } else { if (err != 0) { ksft_print_msg("Spurious event generated for %s\n", ctl->name); + show_values(ctl, initial_val, read_val); ctl->event_spurious++; } } @@ -755,7 +812,6 @@ static bool test_ctl_write_valid_enumerated(struct ctl_data *ctl) static void test_ctl_write_valid(struct ctl_data *ctl) { bool pass; - int err; /* If the control is turned off let's be polite */ if (snd_ctl_elem_info_is_inactive(ctl->info)) { @@ -797,9 +853,7 @@ static void test_ctl_write_valid(struct ctl_data *ctl) } /* Restore the default value to minimise disruption */ - err = write_and_verify(ctl, ctl->def_val, NULL); - if (err < 0) - pass = false; + write_and_verify(ctl, ctl->def_val, NULL); ksft_test_result(pass, "write_valid.%d.%d\n", ctl->card->card, ctl->elem); @@ -1015,9 +1069,7 @@ static void test_ctl_write_invalid(struct ctl_data *ctl) } /* Restore the default value to minimise disruption */ - err = write_and_verify(ctl, ctl->def_val, NULL); - if (err < 0) - pass = false; + write_and_verify(ctl, ctl->def_val, NULL); ksft_test_result(pass, "write_invalid.%d.%d\n", ctl->card->card, ctl->elem); diff --git a/tools/testing/selftests/alsa/pcm-test.c b/tools/testing/selftests/alsa/pcm-test.c index 58b525a4a32c..3e390fe67eb9 100644 --- a/tools/testing/selftests/alsa/pcm-test.c +++ b/tools/testing/selftests/alsa/pcm-test.c @@ -149,6 +149,7 @@ static void missing_devices(int card, snd_config_t *card_config) static void find_pcms(void) { char name[32], key[64]; + char *card_name, *card_longname; int card, dev, subdev, count, direction, err; snd_pcm_stream_t stream; struct pcm_data *pcm_data; @@ -175,6 +176,15 @@ static void find_pcms(void) goto next_card; } + err = snd_card_get_name(card, &card_name); + if (err != 0) + card_name = "Unknown"; + err = snd_card_get_longname(card, &card_longname); + if (err != 0) + card_longname = "Unknown"; + ksft_print_msg("Card %d - %s (%s)\n", card, + card_name, card_longname); + card_config = conf_by_card(card); card_data = calloc(1, sizeof(*card_data)); @@ -489,17 +499,18 @@ __close: } if (!skip) - ksft_test_result(pass, "%s.%s.%d.%d.%d.%s%s%s\n", + ksft_test_result(pass, "%s.%s.%d.%d.%d.%s\n", test_class_name, test_name, data->card, data->device, data->subdevice, - snd_pcm_stream_name(data->stream), - msg[0] ? " " : "", msg); + snd_pcm_stream_name(data->stream)); else - ksft_test_result_skip("%s.%s.%d.%d.%d.%s%s%s\n", + ksft_test_result_skip("%s.%s.%d.%d.%d.%s\n", test_class_name, test_name, data->card, data->device, data->subdevice, - snd_pcm_stream_name(data->stream), - msg[0] ? " " : "", msg); + snd_pcm_stream_name(data->stream)); + + if (msg[0]) + ksft_print_msg("%s\n", msg); pthread_mutex_unlock(&results_lock); |