- 06 1月, 2016 2 次提交
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由 Takashi Sakamoto 提交于
Some models based on ASIC for Dice II series (STD, CP) change their hardware configurations after appearing on IEEE 1394 bus. This is due to interactions of boot loader (RedBoot), firmwares (eCos) and vendor's configurations. This causes current ALSA dice driver to get wrong information about the hardware's capability because its probe function runs just after detecting unit of the model. As long as I investigated, it takes a bit time (less than 1 second) to load the firmware after bootstrap. Just after loaded, the driver can get information about the unit. Then the hardware is initialized according to vendor's configurations. After, the got information becomes wrong. Between bootstrap, firmware loading and post configuration, some bus resets are observed. This commit offloads most processing of probe function into workqueue and schedules the workqueue after successive bus resets. This has an effect to get correct hardware information and avoid involvement to bus reset storm. For code simplicity, this change effects all of Dice-based models, i.e. Dice II, Dice Jr., Dice Mini and Dice III. I use a loose strategy to manage a race condition between the work and the bus reset. This is due to a specification of dice transaction. When bus reset occurs, registered address for the transaction is cleared. Drivers must re-register their own address again. While, this operation is required for the work because the work includes to wait for the transaction. This commit uses no lock primitives for the race condition. Instead, checking 'registered' member of 'struct snd_dice' avoid executing the work again. If sound card is not registered, the work can be scheduled again by bus reset handler. When .remove callback is executed, the sound card is going to be released. The work should not be pending or executed in the releasing. This commit uses cancel_delayed_work_sync() in .remove callback and wait till the pending work finished. After .remove callback, .update callback is not executed, therefore no works are scheduled again. Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Sakamoto 提交于
Before allocating an instance of sound card, ALSA dice driver checks chip_ID_hi in Bus information block of Config ROM, then also checks subaddresses. The former operation reads cache of Config ROM in Linux FireWire subsystem, while the latter operation sends read transaction. The latter can be merged into initialization of transaction system. This commit splits these two operations to reduce needless transactions in probe processing. Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 31 12月, 2015 3 次提交
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由 Julia Lawall 提交于
The dummy_timer_ops structures are never modified, so declare them as const. Done with the help of Coccinelle. Signed-off-by: NJulia Lawall <Julia.Lawall@lip6.fr> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Julia Lawall 提交于
The cs5535audio_dma_ops structures are never modified, so declare them as const. Done with the help of Coccinelle. Signed-off-by: NJulia Lawall <Julia.Lawall@lip6.fr> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Julia Lawall 提交于
The atiixp_dma_ops structures are never modified, so declare them as const. Done with the help of Coccinelle. Signed-off-by: NJulia Lawall <Julia.Lawall@lip6.fr> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 23 12月, 2015 28 次提交
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由 Bard Liao 提交于
There is a status bit on RT5677_PLL1_CTRL2 and RT5677_PLL2_CTRL2. That's why those registers are set volatile. However, the status bit is currently not used by codec driver. So, it should be no problem if we set them non-volatile. The purpose of setting them non-volatile is to restore the setting after a syspend/resume cycle. Signed-off-by: NBard Liao <bardliao@realtek.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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由 Linus Walleij 提交于
As we want gpio_chip .get() calls to be able to return negative error codes and propagate to drivers, we need to go over all drivers and make sure their return values are clamped to [0,1]. We do this by using the ret = !!(val) design pattern. Signed-off-by: NLinus Walleij <linus.walleij@linaro.org> Signed-off-by: NMark Brown <broonie@kernel.org>
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由 Linus Walleij 提交于
As we want gpio_chip .get() calls to be able to return negative error codes and propagate to drivers, we need to go over all drivers and make sure their return values are clamped to [0,1]. We do this by using the ret = !!(val) design pattern. Signed-off-by: NLinus Walleij <linus.walleij@linaro.org> Signed-off-by: NMark Brown <broonie@kernel.org>
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由 Charles Keepax 提交于
The stream is created whilst the compressed stream is opened and a buffer is created when the DSP powers up. It is necessary at a point once both the DSP has powered up and the the stream has been opened to connect a stream to a buffer on the DSP. This is done in the trigger callback as this is after the DSP has been powered and obviously the stream must be open. Note that whilst the connect is currently trivial it is expected that this will get more complex when support for multiple buffers/streams per DSP is added. Signed-off-by: NCharles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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由 Charles Keepax 提交于
Add code that locates and initialises the buffer of compressed data on the DSP if the firmware supported compressed data capture. The buffer struct (wm_adsp_compr_buf) is kept separate from the stream struct (wm_adsp_compr) this will allow much easier support of multiple streams of data from the one DSP in the future, although support for this will not be added in this patch chain. Signed-off-by: NCharles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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由 Charles Keepax 提交于
Allow user-space to open a compressed stream, although no data will be passed yet, as part of this adding the ability to define supported capabilities per firmware and check these match the stream being opened. Signed-off-by: NCharles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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由 Charles Keepax 提交于
Signed-off-by: NCharles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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由 Charles Keepax 提交于
Register a platform driver for the CODEC and add DAIs that will be used to connect a compressed record path for the voice control functionality. Signed-off-by: NCharles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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由 Adam Thomson 提交于
PLL mode based on 32KHz master clock not supported in AB silicon so remove support from the driver. Signed-off-by: NAdam Thomson <Adam.Thomson.Opensource@diasemi.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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由 Adam Thomson 提交于
HW can provide 1.6V micbias level as well the existing levels already provided in the driver. This patch adds support for 1.6V to the DT binding. Signed-off-by: NAdam Thomson <Adam.Thomson.Opensource@diasemi.com> Acked-by: NRob Herring <robh@kernel.org> Signed-off-by: NMark Brown <broonie@kernel.org>
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由 Adam Thomson 提交于
In AB silicon, the internal LDO is not supported so remove DT and driver references to this (digital voltage direct from 'VDD' supply) Signed-off-by: NAdam Thomson <Adam.Thomson.Opensource@diasemi.com> Acked-by: NRob Herring <robh@kernel.org> Signed-off-by: NMark Brown <broonie@kernel.org>
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由 Adam Thomson 提交于
In current AB silicon, BIAS_EN field is enabled by default in the REFERENCES register, so the regmap default value should reflect this. Signed-off-by: NAdam Thomson <Adam.Thomson.Opensource@diasemi.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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由 Adam Thomson 提交于
If codec probe() function fails after supplies have been enabled it should really tidy up and disable them again. This patch updates the probe function to do just that. Signed-off-by: NAdam Thomson <Adam.Thomson.Opensource@diasemi.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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由 Adam Thomson 提交于
Previously Sidetone would operate only when capture to DAI was in progress, due to DAPM path configuration. There is no reason why this should not operate without DAI capture, so this patch updates the DAPM path accordingly. Signed-off-by: NAdam Thomson <Adam.Thomson.Opensource@diasemi.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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由 Maciej S. Szmigiero 提交于
fsl_ssi uses different stream names ("AC97 Playback" / "AC97 Capture") in AC'97 mode so in this case fsl-asoc-card route map should also be using them. Signed-off-by: NMaciej S. Szmigiero <mail@maciej.szmigiero.name> Acked-by: NNicolin Chen <nicoleotsuka@gmail.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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由 Zidan Wang 提交于
Add 8kHz, 11.025kHz, 16kHz, 22.05kHz output sample rate support. According referance menual, "Limited support for the case when output sampling rates is between 8kHz and 30kHz. The limitation is the supported ratio (Fsin/Fsout) range as between 1/24 to 8." Signed-off-by: NZidan Wang <zidan.wang@freescale.com> Acked-by: NNicolin Chen <nicoleotsuka@gmail.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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由 Songjun Wu 提交于
Sometimes the audio play can not be resumed after it is suspended. Add snd_soc_pm_ops to execute power management operations, then this issue is fixed. Signed-off-by: NSongjun Wu <songjun.wu@atmel.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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由 Bard Liao 提交于
Rename some alsa control name as what they should be. Signed-off-by: NBard Liao <bardliao@realtek.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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由 Julia Lawall 提交于
Add NULL test on call to devm_kzalloc. The semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ expression x; identifier fld; @@ * x = devm_kzalloc(...); ... when != x == NULL x->fld // </smpl> Signed-off-by: NJulia Lawall <Julia.Lawall@lip6.fr> Signed-off-by: NMark Brown <broonie@kernel.org>
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由 Julia Lawall 提交于
Add NULL test on call to devm_kzalloc. The semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ expression x; identifier fld; @@ * x = devm_kzalloc(...); ... when != x == NULL x->fld // </smpl> Signed-off-by: NJulia Lawall <Julia.Lawall@lip6.fr> Acked-by: NPeter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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由 Julia Lawall 提交于
Add NULL test on call to devm_kzalloc. The semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ expression x; @@ * x = devm_kzalloc(...); ... when != x == NULL *x // </smpl> Signed-off-by: NJulia Lawall <Julia.Lawall@lip6.fr> Acked-by: NNicolin Chen <nicoleotsuka@gmail.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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由 Markus Elfring 提交于
The return type "unsigned int" was used by the ssm2518_lookup_mcs() function even though it will eventually return a negative error code. Improve this implementation detail by deletion of the type modifier then. This issue was detected by using the Coccinelle software. Signed-off-by: NMarkus Elfring <elfring@users.sourceforge.net> Signed-off-by: NMark Brown <broonie@kernel.org>
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由 Kuninori Morimoto 提交于
This patch adds Multi channel support on Renesas R-Car sound. This patch is tested on Salvator-X board, but it can't use Multi channel, because supported format is different between codec chip and R-Car. Thus, it was tested on board which doesn't mount codec chip, with oscilloscope. Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: NRob Herring <robh@kernel.org> Signed-off-by: NMark Brown <broonie@kernel.org>
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由 Adam Thomson 提交于
This patch adds support to the codec driver to handle mic level detect related IRQs, and report these to user-space using a uevent variable. The uevent variable string "EVENT=MIC_LEVEL_DETECT" is sent to user-space, if the mic level detect feature is enabled, and the audio captured at the chosen mic(s) is above a certain threshold. User-space can then handle the event accordingly (e.g. process audio capture stream). This method was chosen over ALSA control notification for a couple of reasons: 1) There's no requirement here for a control to read state from. The event is the only thing that's required and of interest. 2) tinyalsa support for control notifications does not exist so on platforms using this over alsa-lib there is a need to add code to support this event handling. Another possible option would be to use the standard Jack reporting framework but this really does not fit for this kind of event. Finally, use of the input device framework is not being encouraged, due to difficulties in enabling apps to access input devices, so this has also been avoided. Signed-off-by: NAdam Thomson <Adam.Thomson.Opensource@diasemi.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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由 Hans de Goede 提交于
An external amp (if any) is connected to the external outputs of the SoC of course, rather then directly to the internal amp. Signed-off-by: NHans de Goede <hdegoede@redhat.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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由 Caesar Wang 提交于
Add a device tree match table. This serves to make the driver's support of device tree more explicit. Signed-off-by: NCaesar Wang <wxt@rock-chips.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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由 Koro Chen 提交于
AFE is actually allowed to be turn on before configuration of DAIs since each DAI has its own enabling control. Turn on/off AFE in runtime resume/suspend to avoid AFE being shut down when closing a DAI while other DAIs are still active. Signed-off-by: NKoro Chen <koro.chen@mediatek.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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由 Axel Lin 提交于
Signed-off-by: NAxel Lin <axel.lin@ingics.com> Acked-by: NBard Liao <bardliao@realtek.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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- 22 12月, 2015 7 次提交
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由 Takashi Sakamoto 提交于
As long as I investigate SCS.1m, this model reports to transfer/receive PCM data channels/MIDI conformant data channels in tx/rx AMDTP packet. There's a contradiction that this model actually has no analog/digital capture port for PCM frames and no physical MIDI ports. I guess that SCS.1d also has the contradiction. This model has no analog/digital ports for PCM frames and no physical MIDI ports, thus it requires no streaming functionality. This commit adds some modification codes to handle the contradiction, as much as possible. Unfortunately, this module adds one PCM playback substream for SCS.1d so as SCS.1m. Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Sakamoto 提交于
Now ALSA oxfw driver gains functionalities which scs1x module has. This commit obsoletes the scs1x module, and adds a line of MODULE_ALIAS to load oxfw module instead of scs1x module. In scs1x module, the name of 'shortname' field is fixed as 'SCS1x'. This field is used to name MIDI ports for both of SCS.1m and SCS.1d. This is not good because typically some SCS.1m and SCS.1d are used in the same system. It's better to distinguish them according to name of the ports. This commit applies model name in config ROM to the 'shortname'. For the name of 'driver' and 'longname', this commit uses the same way applied to the other models. This change may not bring disadvantages to users because userspace applications use ALSA rawmidi or seq interface and these interfaces are not influenced by them directly. Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Sakamoto 提交于
This commit adds MIDI playback ports so that scs1x driver has. Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Sakamoto 提交于
This commit copies some functions of asynchronous transactions for MIDI playback, to merge scs1x module. The features of payload in asynchronous transaction are the same as captured MIDI messages. Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Sakamoto 提交于
This commit adds MIDI capture so that scs1x driver has. Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Sakamoto 提交于
This commit copies some functions of asynchronous transactions for MIDI capture, to merge scs1x module. The features of payload in asynchronous transaction are: * System exclusive messages for SCS.1 are encoded without ID data. In this encoding scheme, 4 bits in LSB are available. The bits are squashed in payload byte. Thus, one payload byte transfers two MIDI messages. * The first byte of payload byte means: * 0x00: depending on second payload byte * 0xf9: including escaped system exclusive messages for SCS.1, up to 3 byte (= 6 MIDI messages) * the others: including MIDI 1.0 messages * the others: including escaped system exclusive messages for SCS.1, up to 64 bytes Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Sakamoto 提交于
When physical controls on SCS.1 models are operated, the models transfer MIDI messages in asynchronous transactions on IEEE 1394 bus. The models have a register to have an address for the transactions, and drivers can register own address for this purpose. This commit keeps a region of address, registers it and adds a handler for the transactions. Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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