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2017-04-11Merge branch 'topic/jack' of ↵Mark Brown
git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-rt5665
2017-04-11ALSA: firewire-motu: add tracepoints for messages for unique protocolTakashi Sakamoto
MOTU units transfer/receive messages in each data block of their isochronous packet payload. A part of content in the message is cleard for MIDI message transmission, while the rest is unknown yet. Additional features are required to assist users and developers to reveal the details. This commit adds tracepoints for the purpose. The tracepoints are designed for MOTU's protocol version 2 and 3 (Protocol version 1 is not upstreamed yet). In the tracepoints, events are probed to gather first two 24 bit data chunks of each data block. The chunks are formatted into elements of 64 bit array with padding in MSB. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-11ALSA: firewire-motu: add tracepoints for SPH in IEC 61883-1 fashionTakashi Sakamoto
Unique protocol is used for MOTU FireWire series. In this protocol, data block format is not compliant to AM824 in IEC 61883-1/6. Each of the data block consists of 24 bit data chunks, except for a first quadlet. The quadlet is used for source packet header (SPH) described in IEC 61883-1. The sequence of SPH seems to represent presentation timestamp corresponding to included data. Developers have experienced that invalid sequence brings disorder of units in the series. Unfortunately, current implementation of ALSA IEC 61883-1/6 engine and firewire-motu driver brings periodical noises to the units at sampling transmission frequency based on 44.1 kHz. The engine generates the SPH with even interval and this mechanism seems not to be suitable to the units. Further work is required for this issue and infrastructure is preferable to assist the work. This commit adds tracepoints for the purpose. In the tracepoints, events are probed to gather the SPHs from each data blocks. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-11ALSA: firewire_lib: add tracepoints for packets without CIP headersTakashi Sakamoto
Unique protocol is used for RME Fireface series. In this protocol, payload format for isochronous packet is not compliant to CIP in IEC 61883-1/6. The packet includes data blocks just with data channels, without headers and any metadata. In previous commits, ALSA IEC 61883-1/6 engine supports this protocol. However, tracepoints are not supported yet, unlike implementation for IEC 61883-1/6 protocol. This commit adds support of tracepoints for the protocol. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-10ASoC: sgtl5000: Allow LRCLK pad drive strength to be changedFabio Estevam
Introduce the "lrclk-strength" property to allow LRCLK pad drive strength to be changed via device tree. When running a stress playback loop test on a mx6dl wandboard channel swap can be noticed on about 10% of the times. While debugging this issue I noticed that when probing the SGTL5000 LRCLK pin with the scope the swap did not happen. After removing the probe the swap started to happen again. After changing the LRCLK pad drive strength to the maximum value the issue is gone. Same fix works on a mx6dl Colibri board as well. Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com> Tested-by: Max Krummenacher <max.krummenacher@toradex.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-10ASoC: jack: add snd_soc_codec_set_jackBard Liao
There are many codecs with the capability of jack detection. Usually, we create a jack on machine driver but there is no common function for machine driver to deliver the jack pointer to codec driver. snd_soc_codec_set_jack can be used for delivering the jack pointer to codec driver and enable the jack detection function. To make it work, codec driver need to define a callback function to receive the jack pointer and do all necessary procedure for enabling jack detection. Signed-off-by: Bard Liao <bardliao@realtek.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-10ASoC: hisilicon: Add hi6210 i2s audio driverAndy Green
Add driver for hi6210 i2s controller found on hi6220 boards. Signed-off-by: Andy Green <andy.green@linaro.org> [jstultz: Forward ported to mainline, fairly major rework based on suggestions from Mark Brown] Signed-off-by: John Stultz <john.stultz@linaro.org> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-10ASoC: fsl_ssi: Use the tolower() functionFabio Estevam
Code can be simplified by using the standard tolower() funtion. Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com> Acked-by: Timur Tabi <timur@tabi.org> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-10ASoC: fsl_ssi: Remove FSLSSI_I2S_RATES definitionFabio Estevam
The comment for the FSLSSI_I2S_RATES definition states that the driver currently only supports I2S slave mode, which is no longer correct. As FSLSSI_I2S_RATES is the same as the standard SNDRV_PCM_RATE_CONTINUOUS, just remove its definition and its comments to make the code simpler. Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com> Acked-by: Timur Tabi <timur@tabi.org> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-10ALSA: hda - Allow to enable/disable vmaster build explicitlyTakashi Iwai
Another preliminary patch for the dual-codec support: since the support of vmaster over multiple codecs is difficult, simply disable it by a new flag to hda_codec struct. A new user hint is added as well for consistency. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-10ALSA: hda - A new flag to enforce prefix to each pinTakashi Iwai
This is a preliminary patch for a smooth multi-codec support, and it introduces a new flag, force_pin_prefix, to struct hda_codec. This flag is used to force to add the pin location prefix to each input pin. For example, when there is only one microphone pin, usually the auto-parser assigns the string "Mic". With this flag on, it'll be like "Front Mic". Also, the creation of "Master" or "PCM" playback volume for a single pin is suppressed, too. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=195305 Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-10Merge branch 'for-linus' into for-nextTakashi Iwai
2017-04-07ALSA: hda - set intel audio clock to a proper valueLibin Yang
On some Intel platforms, the audio clock may not be set correctly with initial setting. This will cause the audio playback/capture rates wrong. This patch checks the audio clock setting and will set it to a proper value if it is set incorrectly. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=188411 Signed-off-by: Libin Yang <libin.yang@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-07ALSA: hda - add more ML register definitionsLibin Yang
This patch refines the definition of AZX_MLCTL_SPA and AZX_MLCTL_CPA and add more definitions of ML registers Signed-off-by: Libin Yang <libin.yang@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-06ASoC: STI: Fix null ptr deference in IRQ handlerArnaud Pouliquen
With RTlinux a race condition has been found that leads to NULL ptr crash: - On CPU 0: uni_player_irq_handler is called to treat XRUN "(player->state == UNIPERIF_STATE_STOPPED)" is FALSE so status is checked, dev_err(player->dev, "FIFO underflow error detected") is printed and then snd_pcm_stream_lock should be called to lock stream for stopping. - On CPU 1: application stop and close the stream. Issue is that the stop and shutdown functions are executed while "FIFO underflow error detected" is printed. So when CPU 0 calls snd_pcm_stream_lock, player->substream is already null. Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen@st.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-06ASoC: cs35l35: Correct handling of PDN_DONE with external boostCharles Keepax
When using an external boost supply the PDN_DONE bit is not set, update the handling in this case to use to use an appropriate fixed delay. Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Acked-by: Brian Austin <brian.austin@cirrus.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-06ASoC: cs35l35: Improve power down timeCharles Keepax
Shorten the time it takes to power down the amp by disabling the volume ramp whilst doing the final shutdown. The driver has already muted the amplifier at this stage so doing the volume ramp serves no purpose. Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Acked-by: Brian Austin <brian.austin@cirrus.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-06ASoC: codec: wm8960: Stop when a matching PLL freq is foundDaniel Baluta
When a matching PLL freq is found, searching continues even this is not necessary. The problem was introduced with the following refactoring commit 84fdc00d519ffd ("ASoC: codec: wm9860: Refactor PLL out freq search) Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-06ASoC: Add support for Maxim Integrated MAX98927 AmplifierRyan Lee
Signed-off-by: Ryan Lee <ryans.lee@maximintegrated.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-06ASoC: rsnd: merge rsnd_kctrl_new_m/s/e into rsnd_kctrl_new()Kuninori Morimoto
Current rsnd driver is using rsnd_kctrl_new_m/s/e function, but the differences are very few. This patch merge these rsnd_kctrl_new_m/s/e into rsnd_kctrl_new Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-06Merge branch 'fix/rcar' of ↵Mark Brown
git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-rcar
2017-04-06ASoC: rsnd: tidyup src->convert_rate reset timingKuninori Morimoto
Current src->convert_rate will be set on .hw_param, and be reset on .quit timing. But, .hw_param will not be called again if user did Ctrl-Z + fg. It should be reset on initial of .hw_param to keep its value. Here, ctu.c already do this. This patch solves this issue, other wise, MIXed sound will be strange if user did like below. > aplay -D plughw:0,0 sound_44100.wav & > aplay -D plughw:0,1 sound_96000.wav > Ctrl-Z > fg # 96kHz will be played as 44.1kHz Reported-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-05ALSA: firewire-lib/bebob/oxfw: improve response evaluation for AV/C commandsTakashi Sakamoto
In ALSA firewire stack, some AV/C commands are supported, including vendor's extensions. Drivers includes response parser of each command, according to its requirements, while the parser is written with loose fashion in two points; error check and length check. This doesn't cause any issues such as kernel corruption, but should be improved. This commit modifies evaluations of return value on each parsers. 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>
2017-04-05ALSA: firewire-motu: remove invalid bitshift for register valueTakashi Sakamoto
In protocol version 3, drivers can read current sampling clock status from register 0x'ffff'f000'0b14. 8 bits of LSB of this register represents type of signal as source of clock. Current driver code includes invalid bitshift to handle the parameter. This commit fixes the bug. Reported-by: Dan Carpenter <dan.carpenter@oracle.com> Fixes: 5992e30034c4 ("ALSA: firewire-motu: add support for MOTU 828mk3 (FireWire/Hybrid) as a model with protocol version 3") Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-05ALSA: oxfw: fix regression to handle Stanton SCS.1m/1dTakashi Sakamoto
At a commit 6c29230e2a5f ("ALSA: oxfw: delayed registration of sound card"), ALSA oxfw driver fails to handle SCS.1m/1d, due to -EBUSY at a call of snd_card_register(). The cause is that the driver manages to register two rawmidi instances with the same device number 0. This is a regression introduced since kernel 4.7. This commit fixes the regression, by fixing up device property after discovering stream formats. Fixes: 6c29230e2a5f ("ALSA: oxfw: delayed registration of sound card") Cc: <stable@vger.kernel.org> # 4.7+ Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-05ALSA: firewire-digi00x: remove transaction handler for unknown purposeTakashi Sakamoto
For digi00x series, asynchronous transaction is not used to transfer MIDI messages to/from control surface. One of transction handlers in my previous work loses its practical meaning. This commit removes the handler. I note that unit of console type transfers 0x00001000 to registered address of host space when switching to 'standalone' mode. Then the unit generates bus reset. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-05ALSA: firewire-digi00x: allow user space applications to read/write MIDI ↵Takashi Sakamoto
messages for all ports At a commit c5fcee0373b3 ("ALSA: firewire-digi00x: add MIDI operations for MIDI control port"), I described that MIDI messages for control surface is transferred by a different way from the messages for physical ports. However, this is wrong. MIDI messages to/from all of MIDI ports are transferred by isochronous packets. This commit removes codes to transfer MIDI messages via asynchronous transaction, from MIDI handling layer. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-05ALSA: firewire-digi00x: handle all MIDI messages on streaming packetsTakashi Sakamoto
At a commit 9dc5d31cdceb ("ALSA: firewire-digi00x: handle MIDI messages in isochronous packets"), a functionality to handle MIDI messages on isochronous packet was supported. But this includes some of my misunderstanding. This commit is to fix them. For digi00x series, first data channel of data blocks in rx/tx packet includes MIDI messages. The data channel has 0x80 in 8 bit of its MSB, however it's against IEC 61883-6. Unique data format is applied: - Upper 4 bits of LSB represent port number. - 0x0: port 1. - 0x2: port 2. - 0xe: console port. - Lower 4 bits of LSB represent the number of included MIDI message bytes; 0x0/0x1/0x2. - Two bytes of middle of this data channel have MIDI bytes. Especially, MIDI messages from/to console surface are also transferred by isochronous packets, as well as physical MIDI ports. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-05ALSA: firewire-digi00x: add support for console models of Digi00x seriesTakashi Sakamoto
Digi00x series includes two types of unit; rack and console. As long as reading information on config rom of Digi 002 console, 'MODEL_ID' field has a different value from the one on Digi 002 rack. We've already got a test report from users with Digi 003 rack. We can assume that console type and rack type has different value in the field. This commit adds a device entry for console type. For following commits, this commit also adds a member to 'struct snd_digi00x' to identify console type. $ cd linux-firewire-utils/src $ python2 ./crpp < /sys/bus/firewire/devices/fw1/config_rom ROM header and bus information block ----------------------------------------------------------------- 400 0404f9d0 bus_info_length 4, crc_length 4, crc 63952 404 31333934 bus_name "1394" 408 60647002 irmc 0, cmc 1, isc 1, bmc 0, cyc_clk_acc 100, max_rec 7 (256) 40c 00a07e00 company_id 00a07e | 410 00a30000 device_id 0000a30000 | EUI-64 00a07e0000a30000 root directory ----------------------------------------------------------------- 414 00058a39 directory_length 5, crc 35385 418 0c0043a0 node capabilities 41c 04000001 hardware version 420 0300a07e vendor 424 81000007 --> descriptor leaf at 440 428 d1000001 --> unit directory at 42c unit directory at 42c ----------------------------------------------------------------- 42c 00046674 directory_length 4, crc 26228 430 120000a3 specifier id 434 13000001 version 438 17000001 model 43c 81000007 --> descriptor leaf at 458 descriptor leaf at 440 ----------------------------------------------------------------- 440 00055913 leaf_length 5, crc 22803 444 000050f2 descriptor_type 00, specifier_ID 50f2 448 80000000 44c 44696769 450 64657369 454 676e0000 descriptor leaf at 458 ----------------------------------------------------------------- 458 0004a6fd leaf_length 4, crc 42749 45c 00000000 textual descriptor 460 00000000 minimal ASCII 464 44696769 "Digi" 468 20303032 " 002" Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-05ALSA: fireface: add support for Fireface 400Takashi Sakamoto
Fireface 400 is a second model of RME Fireface series, released in 2006. This commit adds support for this model. This model supports 8 analog channels, 2 S/PDIF channels and 8 ADAT channels in both of tx/rx packet. The number of ADAT channels differs depending on each mode of sampling transmission frequency. $ python2 linux-firewire-utils/src/crpp < /sys/bus/firewire/devices/fw1/config_rom ROM header and bus information block ----------------------------------------------------------------- 400 04107768 bus_info_length 4, crc_length 16, crc 30568 (should be 61311) 404 31333934 bus_name "1394" 408 20009002 irmc 0, cmc 0, isc 1, bmc 0, cyc_clk_acc 0, max_rec 9 (1024) 40c 000a3501 company_id 000a35 | 410 1bd0862a device_id 011bd0862a | EUI-64 000a35011bd0862a root directory ----------------------------------------------------------------- 414 000485ec directory_length 4, crc 34284 418 03000a35 vendor 41c 0c0083c0 node capabilities per IEEE 1394 420 8d000006 --> eui-64 leaf at 438 424 d1000001 --> unit directory at 428 unit directory at 428 ----------------------------------------------------------------- 428 000314c4 directory_length 3, crc 5316 42c 12000a35 specifier id 430 13000002 version 434 17101800 model eui-64 leaf at 438 ----------------------------------------------------------------- 438 000261a8 leaf_length 2, crc 25000 43c 000a3501 company_id 000a35 | 440 1bd0862a device_id 011bd0862a | EUI-64 000a35011bd0862a Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-05ALSA: fireface: add hwdep interfaceTakashi Sakamoto
This commit adds hwdep interface so as the other drivers for audio and music units on IEEE 1394 have. This interface is designed for mixer/control applications. 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> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-05ALSA: fireface: add support for PCM functionalityTakashi Sakamoto
This commit adds PCM functionality to transmit/receive PCM frames on isochronous packet streaming. This commit enables userspace applications to start/stop packet streaming via ALSA PCM interface. Sampling rate requested by applications is used as sampling transmission frequency of IEC 61883-1/6packet streaming. As I described in followed commits, units in this series manages sampling clock frequency independently of sampling transmission frequency, and they supports resampling between their packet streaming/data block processing layer and sampling data processing layer. This commit take this driver to utilize these features for usability. When internal clock is selected as source signal of sampling clock, this driver allows user space applications to start PCM substreams at any rate which packet streaming engine supports as sampling transmission frequency. In this case, this driver expects units to perform resampling PCM frames for rx/tx packets when sampling clock frequency and sampling transmission frequency are mismatched. This is for daily use cases. When any external clock is selected as the source signal, this driver gets configured sampling rate from units, then restricts available sampling rate to the rate for PCM applications. This is for studio use cases. Models in this series supports 64.0/128.0 kHz of sampling rate, however these frequencies are not supported by IEC 61883-6 as sampling transmission frequency. Therefore, packet streaming engine of ALSA firewire stack can't handle them. When units are configured to use any external clock as source signal of sampling clock and one of these unsupported rate is configured as rate of the sampling clock, this driver returns EIO to user space applications. Anyway, this driver doesn't voluntarily configure parameters of sampling clock. It's better for users to work with appropriate user space implementations to configure the parameters in advance of usage. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-05ALSA: fireface: add stream management functionalityTakashi Sakamoto
This commit adds management functionality for packet streaming. As long as investigating Fireface 400, there're three modes depending on sampling transmission frequency. The number of data channels in each data block is different depending on the mode. The set of available data channels for each mode might be different for each protocol and model. The length of registers for the number of isochronous channel is just three bits, therefore 0-7ch are available. When bus reset occurs on IEEE 1394 bus, the device discontinues to transmit packets. This commit aborts PCM substreams at bus reset handler. As I described in followed commits, The device manages its sampling clock independently of sampling transmission frequency against IEC 61883-6. Thus, it's a lower cost to change the sampling transmission frequency, while data fetch between streaming layer and DSP require larger buffer for resampling. As a result, device latency might tend to be larger than ASICs for IEC 61883-1/6 such as DM1000/DM1100/DM1500 (BeBoB), DiceII/TCD2210/TCD2220/TCD3070 and OXFW970/971. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-05ALSA: fireface: add unique data processing layerTakashi Sakamoto
As long as investigating Fireface 400, format of payload of each isochronous packet is not IEC 61883-1/6, thus its format of data block is not AM824. The remarkable points of the format are: * The payload just consists of some data channels of quadlet size without CIP header. * Each data channels includes data aligned to little endian order. * One data channel consists of two parts; 8 bit ancillary field and 24 bit PCM frame. Due to lack of CIP headers, rx/tx packets include no CIP headers and different way to check packet discontinuity. For tx packet, the ancillary field is used for counter. However, the way of counting is different depending on positions of data channels. At 44.1 kHz, ancillary field in: * 1st/6th/9th/10th/14th/17th data channels: not used for this purpose. * 2nd/18th data channels: incremented every data block (0x00-0xff). * 3rd/4th/5th/11th/12th/13th data channels: incremented every 256 data blocks (0x00-0x07). * 7th/8th/15th/16th data channels: incremented per the number of data blocks in a packet. The increment can occur per packet (0x00-0xff). For tx packet, tag of each isochronous packet is used for this purpose. The value of tag cyclically changes between 0, 1, 2 and 3 in this order. The interval is different depending on sampling transmission frequency. At 44.1/48.0 kHz, it's 256 data blocks. At 88.2 kHz, it's 96 data blocks. The number of data blocks in tx packet is exactly the same as SYT_INTERVAL. There's no empty packet or no-data packet, thus the throughput is not 8,000 packets per sec. On the other hand, the one in rx packet is 8,000 packets per sec, thus the number of data blocks is different between each packet, depending on sampling transmission frequency: * 44.1 kHz: 5 or 6 * 48.0 kHz: 5 or 6 or 7 * 88.2 kHz: 10 or 11 or 12 This commit adds data processing layer to satisfy the above specification in a policy of 'best effort'. Although PCM frames are handled for intermediate buffer to user space, the ancillary data is not handled at all to reduce CPU usage, thus counter is not checked. 0 is always used for tag of isochronous packet. Furthermore, the packet streaming layer is responsible for calculation of the number of data blocks for each packet, thus it's not exactly the same sequence from the above observation. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-05ALSA: firewire-lib: add no-header packet processingTakashi Sakamoto
As long as investigating Fireface 400, IEC 61883-1/6 is not applied to its packet streaming protocol. Remarks of the specific protocol are: * Each packet doesn't include CIP headers. * 64,0 and 128,0 kHz are supported. * The device doesn't necessarily transmit 8,000 packets per second. * 0, 1, 2, 3 are used as tag for rx isochronous packets, however 0 is used for tx isochronous packets. On the other hand, there's a common feature. The number of data blocks transferred in a second is the same as sampling transmission frequency. Current ALSA IEC 61883-1/6 engine already has a method to calculate it and this driver can utilize it for rx packets, as well as tx packets. This commit adds support for the transferring protocol. CIP_NO_HEADERS flag is newly added. When this flag is set: * Both of 0 (without CIP header) and 1 (with CIP header) are used as tag to handle incoming isochronous packet. * 0 (without CIP header) is used as tag to transfer outgoing isochronous packet. * Skip CIP header evaluation. * Use unique way to calculate the quadlets of isochronous packet payload. In ALSA PCM interface, 128.0 kHz is not supported, and the ALSA IEC 61883-1/6 engine doesn't support 64.0 kHz. These modes are dropped. The sequence of rx packet has a remarkable quirk about tag. This will be described in later commits. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-05ALSA: firewire-lib: use the same prototype for functions to handle packetTakashi Sakamoto
Audio and music units of RME Fireface series use its own protocol for isochronous packets to transfer data. This protocol requires ALSA IEC 61883-1/6 engine to have alternative functions. This commit is a preparation for the protocol. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-05ALSA: fireface: add proc node to help debuggingTakashi Sakamoto
Drivers can retrieve the state and configuration of clock by read transactions. This commit allows protocol abstraction layer to to dump the information for debugging, via proc interface. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-05ALSA: fireface: add support for MIDI functionalityTakashi Sakamoto
In previous commit, fireface driver supports unique transaction mechanism for MIDI feature. This commit adds MIDI functionality for userspace applications. As I wrote in a followed commit, user space applications get some requirement from this driver. It should not touch a register to which units transmit MIDI messages. It should configure a register in which MIDI transmission is controlled. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-05ALSA: fireface: add transaction supportTakashi Sakamoto
As long as investigating Fireface 400, MIDI messages are transferred by asynchronous communication over IEEE 1394 bus. Fireface 400 receives MIDI messages by write transactions to two addresses; 0x'0000'0801'8000 and 0x'0000'0801'9000. Each of two seems to correspond to MIDI port 1 and 2. Fireface 400 transfers MIDI messages by write transactions to certain addresses which configured by drivers. The drivers can decide upper 4 byte of the addresses by write transactions to 0x'0000'0801'03f4. For the rest part of the address, drivers can select from below options: * 0x'0000'0000 * 0x'0000'0080 * 0x'0000'0100 * 0x'0000'0180 Selected options are represented in register 0x'0000'0801'051c as bit flags. Due to this mechanism, drivers are restricted to use addresses on 'Memory space' of IEEE 1222, even if transactions to the address have some side effects. This commit adds transaction support for MIDI messaging, based on my assumption that the similar mechanism is used on the other protocols. To receive asynchronous transactions, the driver allocates a range of address in 'Memory space'. I apply a strategy to use 0x'0000'0000 as lower 4 byte of the address. When getting failure from Linux FireWire subsystem, this driver retries to allocate addresses. Unfortunately, read transaction to address 0x'0000'0801'051c returns zero always, however write transactions have effects to the other features such as status of sampling clock. For this reason, this commit delegates a task to configure this register to user space applications. The applications should set 3rd bit in LSB in little endian order. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-05ALSA: fireface: add an abstraction layer for model-specific protocolsTakashi Sakamoto
As of 2016, RME discontinued its Fireface series, thus it's OK for us to focus on released firmwares to drive known units. As long as investigating Fireface 400 with Windows driver and comparing the result to FFADO implementation, I can see these firmwares have different register assignments. On the other hand, according to manuals of each models, features relevant to packet streaming seem to be common, because GUIs for these models have the same options. It's reasonable to assume an abstraction layer of protocols to communicate to each models. This commit adds the abstraction layer for the protocols. This layer includes some functions to operate common features of models in this series. In IEC 61883-1/6, the sequence of packet can transfer timing information to synchronize receivers to transmitters. Units of each node on IEEE 1394 bus can generate transmitter's timing clock by handling value of SYT field in CIP header with high-precision clock. For audio and music units on IEEE 1394 bus, this recovered clock is designed to used for sampling clock to capture/generate PCM frames on DSP/ADC/DAC. (Actually, in this world, there's no units to implement this specification as is, as long as I know). Fireface series doesn't use this mechanism. Besides, It doesn't use isochronous packet with CIP header. It uses internal crystal unit as its initial sampling clock. When detecting input signals which can be available for sampling clock (e.g. ADAT input), drivers can configure units to use the signals as source of sampling clock. When something goes wrong, e.g. frequency mismatching between the signal and configured value, units fallback to the other detected signals alternatively. When detecting no alternatives, internal crystal unit is used as source of sampling clock. On manual of Fireface 400, this mechanism is described as 'Autosync'. On the units, packet streaming is controlled by write transactions to certain registers. Format of the packet, e.g. the number of data channels in a data block, is also configured by the same manner. For this purpose, .begin_session and .finish_session is added. The remarkable point of this protocol is to allow drivers to configure arbitrary sampling transmission frequency; e.g. 12.345 Hz. As long as I know, there's no actual DAC/ADC chips which support this kind of capability. I think a pair of packet streaming layer and data block processing layer is isolated from sampling data processing layer in a point of governed clock. In short, between these parts, resampling layer exists. Actually, for Fireface 400, write transactions to 0x'0000'8010'051c has an effect to change sampling clock frequency with base frequencies (32.0/44.1/48.0 kHz) and its multipliers (x2/x4), regardless of sampling transmission frequency. For this reason, the abstraction layer doesn't handle parameters for sampling clock. Instead, each implementation of .begin_session is expected to configure sampling transmission frequency. For packet streaming layer, it's enough to get current selection of source signals for the sampling clock and its frequency. In the abstraction layer, when internal crystal is selected, drivers can sets arbitrary sampling frequency, else they should follow configured frequency. For this purpose, .get_clock is added. Drivers are allows to bank up data fetching from a pair of packet streaming/data block processing layer and sampling data processing layer. This feature seems to suppress noises at starting/stopping packet streaming. For this purpose, .switch_fetching_mode is added. As I described in the above, units have remarkable mechanism to manage sampling clock and process sampling data. For debugging purpose, .dump_sync_status and .dump_clock_config are added. I don't have a need to common interface to represent the status and configuration, developers can add actual implementation of the abstraction layer as they like. Unlike PCM frames, MIDI messages are transferred by asynchronous communication over IEEE 1394 bus, thus target addresses are important for this feature. The .midi_high_addr_reg, .midi_rx_port_0_reg and .midi_rx_port_1_reg are for this purpose. I'll describe them in following commit. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-05ALSA: fireface: add model specific structureTakashi Sakamoto
RME Fireface series has several models and their specifications are different. Currently, we find no way to retrieve the specifications from actual devices and need to implement them in this driver. This commit adds a structure to describe model specific data. This structure has an identical name for each unit, and maximum number of data channels in each mode. I'll describe about the mode in following commits. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-05ALSA: fireface: postpone sound card registrationTakashi Sakamoto
Just after appearing on IEEE 1394 bus, this unit generates several bus resets. This is due to loading firmware from on-board flash memory and initialize hardware. It's better to postpone sound card registration. This commit schedules workqueue to process actual probe processing 2 seconds after the last bus-reset. The card instance is kept at unit probe callback and released at card free callback. Therefore, when the actual probe processing fails, the memory block is wasted. This is due to simplify driver implementation. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-05ALSA: fireface: add skeleton for RME Fireface seriesTakashi Sakamoto
This commit adds a new driver for RME Fireface series. This commit just creates/removes card instance according to IEEE 1394 bus event. More functions will be added in following commits. Three types of firmware have released by RME GmbH; for Fireface 400, for Fireface 800 and for UCX/802/UFX. It's reasonable that these models use different protocol for communication. Currently, I've investigated Fireface 400 and nothing others. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-04-05ASoC: soc-core: verify Sound Card normalityKuninori Morimoto
Current ALSA SoC Sound Card basically consists of CPU/Codec/Platform components. If system uses Kernel modules, we can disable these drivers by using rmmod command. In such case, we can't disable CPU/Codec/Platform driver without disabling Sound Card driver. But on the other hand, we can disable these drivers by using unbind command. In such case, we can disable these drivers randomly. In this case, we can create dirty Sound Card which is missing necessary components. (1) If user disabled Sound Card first, but did nothing to other drivers, user can't use Sound because Sound Card is no longer exists. (2) If user disabled CPU/Codec/Platform driver randomly, but did nothing to Sound Card, user still be able to use Sound Card, because dirty Sound Card still exists. In this case, Sound system will be crashed if user started sound playback/capture. But we can't block such random unbind now. To avoid Sound Card crash in (2) case, we need to unregister Sound Card whenever CPU/Codec/Platform component were unregistered. This patch solves this issue. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-05ASoC: codec: wm9860: Refactor PLL out freq searchDaniel Baluta
Add a separate function for deriving (sysclk, lrclk, bclk) when the clock is auto or pll. Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com> Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-05ASoC: rt5677: Add OF device ID tableJavier Martinez Canillas
The driver doesn't have a struct of_device_id table but supported devices are registered via Device Trees. This is working on the assumption that a I2C device registered via OF will always match a legacy I2C device ID and that the MODALIAS reported will always be of the form i2c:<device>. But this could change in the future so the correct approach is to have an OF device ID table if the devices are registered via OF. Before this patch: $ modinfo sound/soc/codecs/snd-soc-rt5677.ko | grep alias alias: i2c:RT5677CE:00 alias: i2c:rt5676 alias: i2c:rt5677 After this patch: $ modinfo sound/soc/codecs/snd-soc-rt5677.ko | grep alias alias: of:N*T*Crealtek,rt5677C* alias: of:N*T*Crealtek,rt5677 alias: i2c:RT5677CE:00 alias: i2c:rt5676 alias: i2c:rt5677 Signed-off-by: Javier Martinez Canillas <javier@osg.samsung.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-05ASoC: wm8978: Add OF device ID tableJavier Martinez Canillas
The driver doesn't have a struct of_device_id table but supported devices are registered via Device Trees. This is working on the assumption that a I2C device registered via OF will always match a legacy I2C device ID and that the MODALIAS reported will always be of the form i2c:<device>. But this could change in the future so the correct approach is to have an OF device ID table if the devices are registered via OF. Before this patch: $ modinfo sound/soc/codecs/snd-soc-wm8978.ko | grep alias alias: i2c:wm8978 After this patch: $ modinfo sound/soc/codecs/snd-soc-wm8978.ko | grep alias alias: i2c:wm8978 alias: of:N*T*Cwlf,wm8978C* alias: of:N*T*Cwlf,wm8978 Signed-off-by: Javier Martinez Canillas <javier@osg.samsung.com> Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-05ASoC: uda1380: Add OF device ID tableJavier Martinez Canillas
The driver doesn't have a struct of_device_id table but supported devices are registered via Device Trees. This is working on the assumption that a I2C device registered via OF will always match a legacy I2C device ID and that the MODALIAS reported will always be of the form i2c:<device>. But this could change in the future so the correct approach is to have an OF device ID table if the devices are registered via OF. Before this patch: $ modinfo sound/soc/codecs/snd-soc-uda1380.ko | grep alias alias: i2c:uda1380 After this patch: $ modinfo sound/soc/codecs/snd-soc-uda1380.ko | grep alias alias: of:N*T*Cnxp,uda1380C* alias: of:N*T*Cnxp,uda1380 alias: i2c:uda1380 Signed-off-by: Javier Martinez Canillas <javier@osg.samsung.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-05ASoC: sta529: Add OF device ID tableJavier Martinez Canillas
The driver doesn't have a struct of_device_id table but supported devices are registered via Device Trees. This is working on the assumption that a I2C device registered via OF will always match a legacy I2C device ID and that the MODALIAS reported will always be of the form i2c:<device>. But this could change in the future so the correct approach is to have an OF device ID table if the devices are registered via OF. Before this patch: $ modinfo sound/soc/codecs/snd-soc-sta529.ko | grep alias alias: i2c:sta529 After this patch: $ modinfo sound/soc/codecs/snd-soc-sta529.ko | grep alias alias: of:N*T*Cst,sta529C* alias: of:N*T*Cst,sta529 alias: i2c:sta529 Signed-off-by: Javier Martinez Canillas <javier@osg.samsung.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-05ASoC: ssm4567: Add OF device ID tableJavier Martinez Canillas
The driver doesn't have a struct of_device_id table but supported devices are registered via Device Trees. This is working on the assumption that a I2C device registered via OF will always match a legacy I2C device ID and that the MODALIAS reported will always be of the form i2c:<device>. But this could change in the future so the correct approach is to have an OF device ID table if the devices are registered via OF. Before this patch: $ modinfo sound/soc/codecs/snd-soc-ssm4567.ko | grep alias alias: acpi*:INT343B:* alias: i2c:ssm4567 After this patch: $ modinfo sound/soc/codecs/snd-soc-ssm4567.ko | grep alias alias: acpi*:INT343B:* alias: of:N*T*Cadi,ssm4567C* alias: of:N*T*Cadi,ssm4567 alias: i2c:ssm4567 Signed-off-by: Javier Martinez Canillas <javier@osg.samsung.com> Signed-off-by: Mark Brown <broonie@kernel.org>