- 22 Apr, 2013 4 commits
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David Henningsson authored
These are being reported as being so noisy at high mic boost levels, so they are unusable in practice. Therefore artificially limit the boosts. BugLink: https://bugs.launchpad.net/bugs/1089795Signed-off-by:
David Henningsson <david.henningsson@canonical.com> Signed-off-by:
Takashi Iwai <tiwai@suse.de>
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Daniel Schürmann authored
Set the timeout for USB control set messages according to the USB 2 spec, using the macros from include/linux/usb.h. The get timout becomes 5000 ms even though it is 500 ms in the spec. This patch is required to run the Hercules RMX2 which needs a timeout of 1240 ms. More notes from author: I still distinguish between set and get but as long both are 5000 ms GCC will remove it anyway. IMHO this is more easy read and there is no need to explain why we use a get timeout for set messages. Signed-off-by:
Daniel Schürmann <daschuer@mixxx.org> Acked-by:
Daniel Mack <zonque@gmail.com> Signed-off-by:
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Takashi Iwai authored
Like the previous patch by Dan, we should clear the data to be returned from certain compress ioctls, namely, snd_compr_get_codec_caps() and snd_compr_get_params(). This time, we can simply replace kmalloc() with kzalloc(). Signed-off-by: -
Dan Carpenter authored
If the ->get_caps() function doesn't clear the buffer then there would stack information leaked to userspace. For example, soc_compr_get_caps() can return success without clearing the buffer. Signed-off-by:
Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by:
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- 21 Apr, 2013 7 commits
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Charles Keepax authored
This patch reworks the writes to use cumulative values thus making the app_pointer unecessary and removing it. Only tested as far as build. Signed-off-by:
Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by:
Richard Fitzgerald <rf@opensource.wolfsonmicro.com> Acked-by:
Vinod Koul <vinod.koul@intel.com> Signed-off-by:
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Charles Keepax authored
Only tested as far as build. Signed-off-by:
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Charles Keepax authored
Previously we just hard coded all streams as playback streams, this patch checks the DAI to see if it is a capture or playback stream. It is worth noting that at this time only unidirectional streams are supported. Signed-off-by:
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Charles Keepax authored
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Charles Keepax authored
The buffer passed to the copy callback should not be const because the copy callback can be used for capture and playback. Signed-off-by:
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Charles Keepax authored
Signed-off-by:
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Charles Keepax authored
The app_pointer is managed locally by the compress core for memory mapped DSPs but for DSPs that are not memory mapped this would have to be manually updated from within the DSP driver itself, which is hardly very idiomatic. This patch switches to using the cumulative values to calculate the available buffer space because these are already gracefully passed out of the DSP driver to the compress core and otherwise should be functionally equivalent. Signed-off-by:
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- 18 Apr, 2013 14 commits
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Takashi Iwai authored
Merge tag 'asoc-v3.10-2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next ASoC: More updates for v3.10 The main additional change here is Lars-Peter's DMA work plus the platform conversions which have been tested - getting this in mainline will make life easier for development after the merge window. These factor a large chunk of code out of the drivers for the platforms using dmaengine, greatly simplifying development.
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Mark Brown authored
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Mark Brown authored
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Mark Brown authored
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Mark Brown authored
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Mark Brown authored
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Lars-Peter Clausen authored
Use the generic dmaengine PCM driver instead of a custom implemention. There is a minor functional change, the ux500 PCM driver did not preallocate the audio buffer, while the generic dmaengine PCM driver will do this. Signed-off-by:
Lars-Peter Clausen <lars@metafoo.de> Acked-by:
Lee Jones <lee.jones@linaro.org> Signed-off-by:
Mark Brown <broonie@opensource.wolfsonmicro.com>
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Daniel Mack authored
Unfortunately, none of the UAC standards provides a way to identify DSD (Direct Stream Digital) formats. Hence, this patch adds a quirks handler to identify USB interfaces that are capable of handling DSD. That quirks handler can augment the already parsed formats bit-field, by any of the new SNDRV_PCM_FMTBIT_DSD_{U8_U16} and setting the dsd_dop flag in the audio format, if the driver should take care for the DOP byte stuffing. The only devices that are known to work with this are the ones with a 'Playback Designs' vendor id. Signed-off-by: -
Daniel Mack authored
There is quite some confusion around the bit-ordering in DSD samples, and no general agreement that defines whether hardware is supposed to expect the oldest sample in the MSB or the LSB of a byte. ALSA will hence set the rule that on the software API layer, bytes always carry the oldest bit in the most significant bit of a byte, and the driver has to translate that at runtime in order to match the hardware layout. This patch adds support for this by adding a boolean flag to the audio format struct. Signed-off-by:
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Daniel Mack authored
In order to provide a compatibility way for pushing DSD samples through ordinary PCM channels, the "DoP open Standard" was invented. See http://www.dsd-guide.com for the official document. The host is required to stuff DSD marker bytes (0x05, 0xfa, alternating) in the MSB of 24 bit wide samples on the bus, in addition to the 16 bits of actual DSD sample payload. To support this, the hardware and software stride logic in the driver has to be tweaked a bit, as we make the userspace believe we're operating on 16 bit samples, while we in fact push one more byte per channel down to the hardware. The DOP runtime information is stored in struct snd_usb_substream, so we can keep track of our state across multiple calls to prepare_playback_urb_dsd_dop(). Signed-off-by:
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Daniel Mack authored
For normal PCM transfer, this change has no effect, as the endpoint's stride is always frame_bits/8. For DSD DOP streams, however, which is added later, the hardware stride differs from the software stride, and the endpoint has the correct information in these cases. Signed-off-by:
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Daniel Mack authored
This patch adds two formats for Direct Stream Digital (DSD), a pulse-density encoding format which is described here: https://en.wikipedia.org/wiki/Direct_Stream_Digital DSD operates on 2.8, 5.6 or 11.2MHz sample rates and as a 1-bit stream. The two new types added by this patch describe streams that are capable of handling DSD samples in DOP format as 8-bit or in 16-bit (or at a x8 or x16 data rate, respectively). DSD itself specifies samples in *bit*, while DOP and ALSA handle them as *bytes*. Hence, a factor of 8 or 16 has to be applied for the sample rare configuration, according to the following table: configured hardware 176.4KHz 352.8kHz 705.6KHz <---- sample rate 8-bit 2.8MHz 5.6MHz 16-bit 2.8Mhz 5.6MHz 11.2MHz `-----------------------------' actual DSD sample rates Signed-off-by:
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Takashi Iwai authored
When pin default configs are overridden via patch option, these are evaluated before fixups are applied. Since some fixups change the whole codec trees and/or add pins dynamically, this sanity check might not pass when pins aren't present at the time the function is called. We may reorder the execution, but an easier fix is simply to disable this sanity check. Signed-off-by: -
Wei Yongjun authored
Fix to return a negative error code from the error handling case instead of 0, as returned elsewhere in this function. Signed-off-by:
Wei Yongjun <yongjun_wei@trendmicro.com.cn> Signed-off-by:
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- 17 Apr, 2013 14 commits
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Takashi Iwai authored
The flag bus->shutdown implies that the control elements might have been already destroyed. When a codec is resumed at this state and tries to call vmaster hook (e.g. in snd_hda_gen_init()), it would refer to a non-existing object, resulting in Oops in the end. This patch just adds a check of the flag in the caller side for avoiding such a crash. Though, the best would be to clear hook->sw_kctl by the destructor of the corresponding ctl element, but vmaster uses its own private_free, it can't be done easily. So let it be for a while. Signed-off-by: -
Dylan Reid authored
The hardware revision of the codec is based at 0x40. Subtract that before convering to ASCII. The same as it is done for 98095. Signed-off-by:
Dylan Reid <dgreid@chromium.org> Signed-off-by:
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Stas Sergeev authored
This patch adds a playback and capture streams to the dummy codec DAI configuration. Most permissive set of sampling rates and formats is used. This patch is needed for playback and capturing on a codec-less systems, as otherwise the PCM device nodes are not even created. Signed-off-by:
Stas Sergeev <stsp@users.sourceforge.net> Signed-off-by:
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Lars-Peter Clausen authored
Use the generic dmaengine PCM driver instead of a custom implementation. Signed-off-by:
Shawn Guo <shawn.guo@linaro.org> Signed-off-by:
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Lars-Peter Clausen authored
This allows us to access the DAI DMA data when we create the PCM. We'll use this when converting imx to generic DMA engine PCM driver. Signed-off-by:
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Lars-Peter Clausen authored
Unfortunately there are still quite a few platforms with a dmaengine driver which do not support reporting the number of bytes left to transfer. If we want to support these platforms in the generic dmaengine PCM driver we have. Signed-off-by:
Stephen Warren <swarren@nvidia.com> Tested-by:
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Lars-Peter Clausen authored
Use the generic dmaengine PCM driver instead of a custom implementation. Signed-off-by:
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Mark Brown authored
Linux 3.9-rc7
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Lars-Peter Clausen authored
Add support for platforms which don't use devicetree yet or have to optionally support a non-devicetree way to request the DMA channel. The patch adds the compat_request_channel and compat_filter_fn callbacks to the snd_dmaengine_pcm_config struct. If the compat_request_channel is implemented it will be used to request the DMA channel. If not dma_request_channel with compat_filter_fn as the filter function will be used to request the channel. The patch also exports the snd_dmaengine_pcm_request_chan() function, since compat platforms will want to use it to request their DMA channel. Signed-off-by:
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Lars-Peter Clausen authored
This patch adds a generic dmaengine PCM driver. It builds on top of the dmaengine PCM library and adds the missing pieces like DMA channel management, buffer management and channel configuration. It will be able to replace the majority of the existing platform specific dmaengine based PCM drivers. Devicetree is used to map the DMA channels to the PCM device. Signed-off-by:
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Lars-Peter Clausen authored
snd_soc_{add,remove}_platform are similar to snd_soc_register_platform and snd_soc_unregister_platform with the difference that they won't allocate and free the snd_soc_platform structure. Also add snd_soc_lookup_platform which looks up a platform by the device it has been registered for. Signed-off-by: -
Mark Brown authored
Merge branch 'topic/core' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-dma
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Lars-Peter Clausen authored
Refactor the dmaengine PCM library to allow the DMA channel to be requested before opening a PCM substream. snd_dmaengine_pcm_open() now expects a DMA channel instead of a filter function and filter parameter as its parameters. snd_dmaengine_pcm_close() is updated to not release the DMA channel. This allows a dmaengine based PCM driver to request its channels before the substream is opened. The patch also introduces two new functions, snd_dmaengine_pcm_open_request_chan() and snd_dmaengine_pcm_close_release_chan(), which have the same signature and behaviour of the old snd_dmaengine_pcm_{open,close}() and internally use the new variants of these functions. All users of snd_dmaengine_pcm_{open,close}() are updated to use snd_dmaengine_pcm_open_request_chan() and snd_dmaengine_pcm_close_release_chan(). Signed-off-by: -
David Henningsson authored
When graphics initializes the HDMI chip, sometimes this leads to pins going into D3 and right channel being muted. If the audio driver finishes initialization before the graphic driver does, this situation becomes permanent. This is a workaround that checks for this situation and corrects it on playback prepare. It has been verified working on at least one machine. BugLink: https://bugs.launchpad.net/bugs/1167270Signed-off-by:
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- 16 Apr, 2013 1 commit
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Takashi Iwai authored
When setting up the aamix output paths, use the primary DAC instead of the individual DAC for each output as default. Otherwise multiple DACs will be turned on for a single aamix widget, which results in doubly or more volumes, because the duplicated signals will be sent through all these DACs for a single stream. Signed-off-by:
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