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* ALSA: bebob: add support for Behringer FCA 610/1616Takashi Sakamoto2015-06-152-2/+5
| | | | | | | | | | | | | | | | | | | | | | | | They're based on DM1500 (ArchWave produced), and BeBoB version 3 is installed. $ cat /proc/asound/FCA610/firewire/firmware Manufacturer: bridgeCo Protocol Ver: 3 Build Ver: 0 GUID: 0x001564000002AD73 Model ID: 0x03 Model Rev: 0 Firmware Date: 20121102 Firmware Time: 153431 Firmware ID: 0x610 Firmware Ver: 8348 Base Addr: 0x400C0080 Max Size: 1422624 Loader Date: 20121015 Loader Time: 104710 Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: bebob: keep duplex streams always to keep internal multiplexer properlyTakashi Sakamoto2015-06-154-27/+16
| | | | | | | | | | | | | | | | | Behringer FCA610 transmits packets with periodic noisy PCM samples when receiving no streams, and generates a bit noisy sound. ALSA BeBoB driver is programmed to establish both in/out connections when starting streaming, then transfers packets as userspace applications requested. This means that there's a case that one of incoming/outgoing streams is running, to save CPU and bandwidth usage. Although, it's natural to start transferring packets in both direction. This commit makes this driver to keeps duplex streams always. Tested-by: Kim Tore Jensen <kim@incendio.no> Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: bebob: loosen up severity of checking continuity for BeBoB v3 quirkTakashi Sakamoto2015-06-151-0/+11
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | PrismSound Orpheus, Behringer UFX1604 and FCA610 work with BeBoB v3, and they're confirmed to transmit discontinuous packets in the beginning of streaming. payload CIP headers 8 0x00070000 0x9002FFFF 8 0x00070000 0x9002FFFF 8 0x00070000 0x9002FFFF 8 0x00070008 0x9002FFFF <- 8 0x00070008 0x9002FFFF 8 0x00070008 0x9002FFFF 8 0x00070008 0x9002FFFF 8 0x00070008 0x9002FFFF 8 0x00070008 0x9002FFFF 232 0x00070000 0x9002E798 <- 232 0x00070008 0x9002FB99 232 0x00070010 0x90021398 8 0x00070018 0x9002FFFF (This sample was got with Behringer FCA610 and FFADO library.) This commit sets CIP_EMPTY_HAS_WRONG_DBC and CIP_SKIP_DBC_ZERO_CHECK to ignore these discontinuities. Tested-by: Kim Tore Jensen <kim@incendio.no> Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: bebob: expand timeout for DM1500 quirkTakashi Sakamoto2015-06-151-1/+1
| | | | | | | | | | | | | | | | | Behringer FCA610 and UFX1604 is confirmed to require more time till transmitting packets after establishing connections. This seems to be a quirk of DM1500 ASIC which ArchWave produced. For this quirk, this commit extends the time to wait up to 2 seconds. As a result, in worst cases, below userspace functions require 2 seconds to return. - snd_pcm_prepare() - snd_pcm_hw_params() - snd_pcm_recover() Tested-by: Kim Tore Jensen <kim@incendio.no> Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: bebob: add 'version' member for BeBoB protocol versionTakashi Sakamoto2015-06-152-0/+11
| | | | | | | | | | | | | BeBoB installed devices have BeBoB register area. This area stores basic information about its firmware. A register has its protocol version. This commit adds 'version' member and store the device's protocol version to handle v3 quirks in following commits. Tested-by: Kim Tore Jensen <kim@incendio.no> Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: bebob: add SYT-Match supportTakashi Sakamoto2015-06-152-7/+18
| | | | | | | | | | In previous commits, this driver can detect the source of clock as mush as possible. SYT-Match mode is also available. This commit purge the restriction. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: bebob: obsolete string literal expression for clock sourceTakashi Sakamoto2015-06-155-27/+6
| | | | | | | | | | The old string literals were completely replaced by new normalized representation. This commit obsoletes it. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: bebob: use normalized representation for the type of clock sourceTakashi Sakamoto2015-06-154-31/+33
| | | | | | | | | This commit changes function prototype and its processing. As a result, function caller can execute additional processing according to detected clock source. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: bebob: preparation for replacing string literals by normalized ↵Takashi Sakamoto2015-06-154-12/+72
| | | | | | | | | | | | | representation for model-dependent structures Previous commit adds a enumerator as a normalized representation of clock source, while model-dependent structures still use string literals for this purpose. This commit is a preparation for replacement. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: bebob: apply new enumerator to normalize the type of clock sourceTakashi Sakamoto2015-06-151-0/+6
| | | | | | | | | | Previous commit allows this driver to detect several types of clock source, while there's no normalized expression for it. This commit adds a new enumerator for this purpose. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: bebob: improve signal mode detection for clock sourceTakashi Sakamoto2015-06-152-9/+72
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | With BeBoB version 3, current ALSA BeBoB driver detects the type of current clock signal source wrongly. This is due to a lack of proper implementation to parse the information. This commit renews the parser. As a result, this driver detects SYT-Match clock signal, thus it can start streams with two modes; SYT-Match mode and the others. SYT-Match mode will be supported in future commits. There's a constrain about detected internal/external clock source. When detecting external clock source, this driver allows userspace applications to use current sampling rate only. This is due to consider abour synchronization to external clock sources such as S/PDIF, ADAT or word-clock. According to several information from some devices, I guesss that the internal clock of most devices synchronize to IEEE 1394 cycle start packet. In this case, by a usual way, it's detect as 'Sync type of output Music Sub-Unit' connected to 'Sync type of PCR output Unit (oPCR)', and this driver judges it as internal clock. Therefore, userspace applications is allowed to request arbitrary supported sampling rates. On the other hand, several devices based on BeBoB version 3 have additional internal clock. In this case, by a usual way, it's detect as 'Sync/Additional type of External input Unit'. Unfortunately, there's no way to distinguish this sync type from the other external clock sources such as word-clock. In this case, this driver handles it as external and userspace applications is forced to use current sampling rate. I note that when the source of clock is detected as 'Isochronous stream type of input PCR[0]', it's under 'SYT-Match' mode. In this mode, the synchronization clock is generated according to SYT-series in received packets. In this case, this driver generates the series by myself. I experienced this mode often make the device silent suddenly during playbacking. This means that the mode is easy to lost synchronization. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: firewire-lib: fix buffer-over-run when detecting packet discontinuityTakashi Sakamoto2015-05-271-16/+16
| | | | | | | | | | | | | | | | | | | | | When detecting packet discontinuity, handle_in_packet() returns minus value and this value is assigned to unsigned int variable, then the variable has huge value. As a result, the variable causes buffer-over-run in handle_out_packet(). This brings invalid page request and system hangup. This commit fixes the bug to add a new argument into handle_in_packet() and the number of handled data blocks is assignd to it. The function return value is just used to check error. I also considered to change the type of local variable to 'int' in in_stream_callback(). This idea is based on type-conversion in C standard, while it may cause future problems when adding more works. Thus, I dropped this idea. Fixes: 6fc6b9ce41c6('ALSA: firewire-lib: pass the number of data blocks in incoming packets to outgoing packets') Reported-by: Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: bebob: add Digidesign Mbox 2 Pro supportTakashi Sakamoto2015-05-242-0/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This device is based on DM1000E, and BeBoB version 1 firmware is installed. $ cat /proc/asound/cards 0 [Pro ]: BeBoB - Mbox 2 Pro DIGIDESIGN Mbox 2 Pro (id:1, rev:1), GUID 00a07e0100a90000 at fw1.0, S400 $ cat /proc/asound/Pro/firewire/firmware Manufacturer: bridgeCo Protocol Ver: 1 Build Ver: 0 GUID: 0x00A07E0100A90000 Model ID: 0x01 Model Rev: 1 Firmware Date: 20071031 Firmware Time: 034402 Firmware ID: 0xA9 Firmware Ver: 16777215 Base Addr: 0x20080000 Max Size: 1572864 Loader Date: 20051207 Loader Time: 205554 With this patch, ALSA BeBoB driver can start packet streaming to/from this model, while as a default, internal multiplexer of this model is not initialized and generates no sound even if the driver transfers any packets with PCM samples. To hear any sounds from this model, userspace applications should be developed to set parameters to the internal multiplexer. You can see raw information in FFADO website: http://subversion.ffado.org/wiki/AvcModels/DigiDesignMboxPro2 Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: firewire-lib: use protocol error when detecting wrong value in CIP headerTakashi Sakamoto2015-05-241-1/+1
| | | | | | | | | | | | | When detecting zero in 'dbs' field of CIP header, this packet streaming should be aborted because of avoiding division-by-zero. This is an error in an aspect of IEC 61883-1, thus protocol error. This commit use EPROTO instead of EIO. Actually, the returned value is not used for userspace and this commit has no effect. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Acked-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: firewire-lib: use dev_err() when detecting incoming streaming errorTakashi Sakamoto2015-05-241-4/+4
| | | | | | | | | | | | | | | When detecting invalid value in 'dbs' field of CIP header or packet discontinuity, current implementation reports the status by err_info(). In most cases this state is caused by model-specific issue due to vendor's customization and should be reported to developers. This commit use dev_err() instead of dev_info() for this purpose. In the cases, packet streaming is aborted, thus no message floading occurs. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Acked-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: firewire-lib: macro arrangement for code cleanupTakashi Sakamoto2015-05-241-14/+20
| | | | | | | | | | | Some macros include my misunderstanding for IEC 61883-1 or -6. Additionally, some fixed values appear on codes. This commit replaces these with macros with proper names. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Acked-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: firewire-lib: rename local functions for code cleanupTakashi Sakamoto2015-05-241-30/+30
| | | | | | | | | The naming rule for local functions was inconsistent. This commit rename them with a consistent manner. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Acked-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: firewire-lib: remove restriction for non-blocking modeTakashi Sakamoto2015-05-231-1/+1
| | | | | | | | Former patches allow non-blocking streams to synchronize with timestamp. This patch removes the restriction. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: firewire-lib: set streaming error outside of packetizationTakashi Sakamoto2015-05-231-15/+24
| | | | | | | | | | In previous commit, error handling for incoming packet processing is outside of packetization. This is nice for reading the codes. This commit applies this idea for outgoing packet processing, too. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: firewire-lib: pass the number of data blocks in incoming packets to ↵Takashi Sakamoto2015-05-231-24/+30
| | | | | | | | | | | | | | | | | outgoing packets Current implementation reuses the value of syt field in incoming packet to outgoing packet for full duplex timestamp synchronization, while the number of data blocks in outgoing packets refers to hard-coded table and the synchronization cannot be applied to non-blocking stream. This commit passes the number of data blocks from incoming packet processing to outgoing packet processing for the synchronization. For normal mode, isochronous callback handler is changed to generate the values of syt and data blocks. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: firewire-lib: simplify function to calculate the number of data blocksTakashi Sakamoto2015-05-231-22/+27
| | | | | | | | | | | This function is called according to conditions between the value of syt and streaming mode(blocking or non-blocking). To simplify caller's work, this commit push these conditions to the function. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: firewire-lib: add buffer-over-run protection at receiving more data ↵Takashi Sakamoto2015-05-233-4/+31
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | blocks than expected In IEC 61883-6, the number of data blocks in a packet is limited up to the value of SYT_INTERVAL. Current implementation is compliant to the limitation, while it can cause buffer-over-run when the value of dbs field in received packet is illegally large. This commit adds a validator to detect such illegal packets to prevent the buffer-over-run. Actually, the buffer is aligned to the size of memory page, thus this issue hardly causes system errors due to the room to page alignment, as long as a few packets includes such jumbo payload; i.e. a packet to several received packets. Here, Behringer F-Control Audio 202 (based on OXFW 960) has a quirk to postpone transferring isochronous packet till finish handling any asynchronous packets. In this case, this model is lazy, transfers no packets according to several cycle-start packets. After finishing, this model pushes required data in next isochronous packet. As a result, the packet include more data blocks than IEC 61883-6 defines. To continue to support this model, this commit adds a new flag to extend the length of calculated payload. This flag allows the size of payload 5 times as large as IEC 61883-6 defines. As a result, packets from this model passed the validator successfully. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Merge branch 'for-next' into for-linusTakashi Iwai2015-04-132-5/+5
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| * Merge branch 'for-linus' into for-nextTakashi Iwai2015-03-123-13/+12
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| * | ALSA: firewire: Fix trivial typos in commentsYannick Guerrini2015-03-102-5/+5
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Change 'propper' to 'proper' Change 'paramters' to 'parameters' Change 'SYT_INTEVAL' to 'SYT_INTERVAL' Change 'aligh'/'alighed' to 'align'/'aligned' Signed-off-by: Yannick Guerrini <yguerrini@tomshardware.fr> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: bebob: fix to processing in big-endian machine for sending cueTakashi Sakamoto2015-04-081-4/+4
| |/ |/| | | | | | | | | | | | | | | | | | | | | | | | | Some M-Audio devices require to receive bootup command just after powering on, while codes in BeBoB driver doesn't work properly in big-endian machine because the command should be aligned by little-endian. This commit fixes this bug. This fix should go to stable kernel. Cc: Takayuki Shiroma <t.shiroma.oki@gmail.com> Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: firewire-lib: leave unit reference counting completelyTakashi Sakamoto2015-03-101-2/+1
| | | | | | | | | | | | | | | | | | | | | | | | With previous commit, this module managed to leave the counting to each drivers, but the isochronous resources functionality still increment/decrement the count. This commit purge such codes to leave the responsibility to each drivers. Fix: c6f224dc20ad ('ALSA: firewire-lib: remove reference counting') Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | Revert "ALSA: dice: fix wrong offsets for Dice interface"Takashi Sakamoto2015-03-102-11/+11
|/ | | | | | | | | This reverts commit 8cdebf71098c07168ef6335e2f1f35d85dbe3049. The reverted commit breaks out-stream functionality of Dice driver. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: dice: fix wrong offsets for Dice interfaceTakashi Sakamoto2015-03-012-11/+11
| | | | | | | | | | | | | | For received packet stream, the offset of 'RX_SEQ_START' locates after the offset of 'RX_NUMBER_MIDI', although current macro and proc output includes wrong offsets. Fortunately, this bug doesn't affect streaming functionality because these macro is not used. This commit fixes these wrong macro and outputs. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: oxfw: fix a condition and return code in start_stream()Takashi Sakamoto2015-02-271-2/+3
| | | | | | | | | | | | | The amdtp_stream_wait_callback() doesn't return minus value and the return code is not for error code. This commit fixes with a propper condition and an error code. Fixes: f3699e2c7745 ('ALSA: oxfw: Change the way to start stream') Reported-by: Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Cc: <stable@vger.kernel.org> # 3.19+ Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: fireworks/bebob/dice/oxfw: make it possible to shutdown safelyTakashi Sakamoto2015-02-236-28/+15
| | | | | | | | | | | | | A part of these drivers, especially BeBoB driver, are programmed to wait some events. Thus the drivers should not destroy any data in .remove() context. This commit moves some destructors from 'struct fw_driver.remove()' to 'struct snd_card.private_free()' to shutdown safely. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Cc: <stable@vger.kernel.org> # 3.19+ Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: fireworks/bebob/dice/oxfw: allow stream destructor after releasing runtimeTakashi Sakamoto2015-02-234-21/+30
| | | | | | | | | | | | | | | | | | | | Currently stream destructor in each driver has a problem to be called in a context in which sound card object is released, because the destructors call amdtp_stream_pcm_abort() and touch PCM runtime data. The PCM runtime data is destroyed in application's context with snd_pcm_close(), on the other hand PCM substream data is destroyed after sound card object is released, in most case after all of ALSA character devices are released. When PCM runtime is destroyed and PCM substream is remained, amdtp_stream_pcm_abort() touches PCM runtime data and causes Null-pointer-dereference. This commit changes stream destructors and allows each driver to call it after releasing runtime. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Cc: <stable@vger.kernel.org> # 3.19+ Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: firewire-lib: remove reference countingTakashi Sakamoto2015-02-231-2/+1
| | | | | | | | | | | | | AMDTP helper functions increment/decrement reference counter for an instance of FireWire unit, while it's complicated for each driver to process error state. In previous commit, each driver has the role of reference counting. This commit removes this role from the helper function. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Cc: <stable@vger.kernel.org> # 3.19+ Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: fireworks/bebob/dice/oxfw: add reference-counting for FireWire unitTakashi Sakamoto2015-02-234-4/+42
| | | | | | | | | | | | | | | | | Fireworks and Dice drivers try to touch instances of FireWire unit after sound card object is released, while references to the unit is decremented in .remove(). When unplugging during streaming, sound card object is released after .remove(), thus Fireworks and Dice drivers causes GPF or Null-pointer-dereferencing to application processes because an instance of FireWire unit was already released. This commit adds reference-counting for FireWire unit in drivers to allow them to touch an instance of FireWire unit after .remove(). In most case, any operations after .remove() may be failed safely. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Cc: <stable@vger.kernel.org> # 3.19+ Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: firewire-lib: fix an unexpected byte sequence for micro signTakashi Sakamoto2015-02-231-1/+1
| | | | | | | | | | | | | | | The sign for microsecond (U+0085, MICRO SIGN) was encoded to '0x c2 b5' by UTF-8 character encoding scheme. But the byte sequence was converted to '0x c3 82 c2 b5' in a previous commit. As a result, the byte sequence cannot represent microsecond sign in UTF-8 or ASCII. This may confuse developers. This commit replaces the sign to string expression with 'microseconds' to purge superfluous troubles. Fixes: 5c697e5b46ef("ALSA: firewire-lib: remove rx_blocks_for_midi quirk") Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: firewire-lib: limit the MIDI data rateClemens Ladisch2015-01-162-6/+57
| | | | | | | | | | | | Do no send MIDI bytes at the full rate at which FireWire packets happen to be sent, but restrict them to the actual rate of a real MIDI port. This is required by the specification, and prevents data loss when the device's buffer overruns. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Tested-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: firewire-lib: remove rx_blocks_for_midi quirkClemens Ladisch2015-01-164-19/+8
| | | | | | | | | | | | | There are several devices that expect to receive MIDI data only in the first eight data blocks of a packet. If the driver restricts the data rate to the allowed rate (as mandated by the specification, but not yet implemented by this driver), this happens naturally. Therefore, there is no reason to ever try to use more data packets with any device. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Tested-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: fireworks: fix an endianness bug for transaction lengthTakashi Sakamoto2015-01-071-1/+1
| | | | | | | | | | Although the 't->length' is a big-endian value, it's used without any conversion. This means that the driver always uses 'length' parameter. Fixes: 555e8a8f7f14("ALSA: fireworks: Add command/response functionality into hwdep interface") Reported-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: oxfw: some signedness bugsDan Carpenter2014-12-153-5/+6
| | | | | | | | | | | | | This code tends to use unsigned variables by default and it causes signedness bugs when we use negative variables for error handling. The "i" and "j" variables are used to iterated over small positive values and so they should be type "int". The "len" variable doesn't *need* to be signed but it should be signed to make the code easier to read and audit. Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com> Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: oxfw: fix detect_loud_models() return valueDan Carpenter2014-12-141-1/+1
| | | | | | | | | | | | | | | This code causes a static checker warning: sound/firewire/oxfw/oxfw.c:46 detect_loud_models() warn: signedness bug returning '(-2)' The detect_loud_models() function should return false on falure, so that we don't try to set up the loud code for hardware that doesn't support it. Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com> Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: oxfw: Add hwdep interfaceTakashi Sakamoto2014-12-108-2/+276
| | | | | | | | | | | This interface is designed for mixer/control application. By using this interface, an application can get information about firewire node, can lock/unlock kernel streaming and can get notification at starting/stopping kernel streaming. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Acked-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: oxfw: Add support for capture/playback MIDI messagesTakashi Sakamoto2014-12-105-1/+208
| | | | | | | | | | | | | | | | | | This commit adds MIDI functionality with an assumption of 'if the device has MIDI comformant data channels in its stream formation, the device has one MIDI port'. When no streams have already started, MIDI functionality starts stream with current sampling rate. When MIDI functionality has already starts some streams and PCM functionality is going to start streams at different sampling rate, this driver stops streams once and changes sampling rate, then restarts streams for both PCM/MIDI substreams. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Acked-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: oxfw: add support for capturing PCM samplesTakashi Sakamoto2014-12-101-20/+180
| | | | | | | | | | | | | In previous commit, a support for transmitted packets is added. This commit add a support for capturing PCM samples. When any streams are already started, this driver should not change sampling rate of the device, thus this commit also adds a restriction of sampling rate in this situation. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Acked-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: oxfw: Add support AMDTP in-streamTakashi Sakamoto2014-12-105-75/+279
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Previous commit adds support for some devices which can capture PCM samples. These devices transmit AMDTP stream in non-blocking mode. This commit adds functionality to handle AMDTP incoming stream. OXFW seems to have two quirks: - Transmits packets with non-zero dbc in its beginning - Transmits packets with wrong values in syt field For the first quirk, this commit adds CIP_SKIP_INIT_DBC_CHECK flag for incoming stream to skip first check of dbc. For the second quirk, this commit doesn't add duplex stream which Fireworks/BeBoB drivers use. So OXFW driver generates syt value for outgoing stream. Here are examples of a sequence of packets transmitted by Behringer F-Control Audio 202. There are differences between sequences of syt value when OXFW driver transfers outgoing stream or not. When driver gives no outgoing stream: Index Payload CIP_Header_0 CIP_Header_1 38 14 00020092 900103D1 39 12 00020098 900102FF 40 12 0002009D 9001027F 41 14 000200A2 90010396 42 14 000200A8 900102E8 43 12 000200AE 90010219 44 14 000200B3 90010331 45 12 000200B9 9001025F 46 14 000200BE 90010376 47 12 000200C4 900102A1 00 12 000200C9 9001023E 01 14 000200CE 90010358 02 12 000200D4 90010289 03 16 000200D9 900103A3 04 12 000200E0 900102DD 05 14 000200E5 900103F1 06 12 000200EB 90010335 07 12 000200F0 90010263 08 14 000200F5 9001037C 09 12 000200FB 900102AE When driver gives outgoing stream: Index Payload CIP_Header_0 CIP_Header_1 38 12 000200BD 900104A8 39 14 000200C2 900104A8 40 12 000200C8 900104AC 41 14 000200CD 900104A9 42 12 000200D3 900104B1 43 14 000200D8 900104A8 44 12 000200DE 900104AA 45 14 000200E3 900104A9 46 14 000200E9 900104AE 47 12 000200EF 900104A8 00 14 000200F4 900104AD 01 12 000200FA 900104A7 02 14 000200FF 900104A9 03 12 00020005 900104A9 04 14 0002000A 900104B1 05 12 00020010 900104AA 06 14 00020015 900104AD 07 12 0002001B 900104A7 08 14 00020020 900104AC 09 12 00020026 900104A7 Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Acked-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: oxfw: Add support for Behringer/Mackie devicesTakashi Sakamoto2014-12-102-6/+91
| | | | | | | | | | | | | | | | | Some devices produced by Behringer/Mackie are based on OXFW970/971. This commit adds support for them. Additionally, this commit changes the way to name card with some information in config rom. Ids of some Mackie(Loud) models are not identified, therefore this commit applies name detection for these models. The devices support capture/playback of PCM-samples and some of them supports capture/playback of MIDI messages. These functionalities are implemented by followed commits. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Acked-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: oxfw: Change the way to start streamTakashi Sakamoto2014-12-103-48/+138
| | | | | | | | | | | | | | | In past commit, this driver can keep stream formations for each sampling rate. So its stream functionality can decide stream formations with given some parameters. This commit moves related codes from PCM functionality to stream functionality. Furthermore, to set stream format correctly, this commit uses AV/C Stream Format Information command instead of AV/C Input/Output Plug Signal Format command. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Acked-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: oxfw: Add proc interface for debugging purposeTakashi Sakamoto2014-12-105-1/+118
| | | | | | | | | | This commit adds proc interface to get information about stream formation. This commit also adds snd_oxfw_stream_get_current_formation() to get current stream formation. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Acked-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: oxfw: Change the way to make PCM rules/constraintsTakashi Sakamoto2014-12-104-86/+410
| | | | | | | | | | | | | | | | | | | In previous commit, this driver can get to know stream formations at each supported sampling rates. This commit uses it to make PCM rules/constraints and obsoletes hard-coded rules/constraints. For this purpose, this commit adds 'struct snd_oxfw_stream_formation' and snd_oxfw_stream_parse_format() to parse data channel formation of data block. According to datasheet of OXFW970/971, they support 32.0kHz to 196.0kHz. As long as developers investigate, some devices are confirmed to have several formats for the same sampling rate. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Acked-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: oxfw: Add support for AV/C stream format command to get/set supported ↵Takashi Sakamoto2014-12-103-1/+187
| | | | | | | | | | | | | | | | | | | | | | | | | | | | stream formation OXFW970/971 may supports AV/C Stream Format Information Specification 1.1 Working Draft (Apr 2005, 1394TA). By using this command, drivers can get to know stream formations which device supports. This commit adds 'EXTENDED STREAM FORMAT INFORMATION' command. This command has two subfunctions, 'SINGLE' and 'LIST'. Drivers can use 'SINGLE' subfunction to know/set current formation of AMDTP stream, Drivers can use 'LIST' subfunction to know an available formation of AMDTP stream in a certain sampling rate. But some devices don't implement the 'LIST' subfunction. So this commit uses an assumption that 'if they don't implement it, they don't change stream formation depending on current each sampling rate'. With this assumption, this driver generates formations for such devices by: 1.getting current formation by SINGLE subfunction 2.getting supported sampling rates 3.applying current formation for all of supported sampling rates Followed commit implements a parser of this format information. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Acked-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: oxfw: Change the way to name cardTakashi Sakamoto2014-12-102-23/+34
| | | | | | | | | | | | | | This is a preparation for more models. In following commit, members of 'struct snd_card' related to name becomes to consists of vendor and model strings in device's config-rom. Current supported devices also has strings in their config rom, but the strings are too long to name sound card, thus this driver still keep hard-coded vendor and model names for them. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Acked-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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