- 13 4月, 2012 2 次提交
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由 Daniel Mack 提交于
Implicit feedback is a streaming mode that does not rely on dedicated sync endpoints but uses the information provided by record streams to clock output streams. Now that the streaming logic is decoupled from the PCM streams, this is easy to implement. Signed-off-by: NDaniel Mack <zonque@gmail.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Daniel Mack 提交于
With the previous commit that added the new streaming model, all endpoint and streaming related code is now in endpoint.c, and pcm.c only acts as a wrapper for handling the packet's payload. Signed-off-by: NDaniel Mack <zonque@gmail.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 15 3月, 2012 1 次提交
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由 Takashi Iwai 提交于
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 27 9月, 2011 1 次提交
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由 Clemens Ladisch 提交于
There are certain devices that are reportedly so slow that they need more than 100 ms to handle control transfers. Therefore, increase the timeout in mixer(_quirks).c to 1000 ms. The timeout parameter of snd_usb_ctl_msg() is now constant, so we can drop it. Reported-by: NFelipe Balbi <balbi@ti.com> Signed-off-by: NClemens Ladisch <clemens@ladisch.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 14 9月, 2011 1 次提交
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由 Daniel Mack 提交于
No code altered at this point, simply preparing for upcoming refactorizations. Signed-off-by: NDaniel Mack <zonque@gmail.com> Acked-by: NClemens Ladisch <clemens@ladisch.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 12 9月, 2011 1 次提交
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由 Pierre-Louis Bossart 提交于
Existing code only updates the audio delay when URBs were submitted/retired. This can introduce an uncertainty of 8ms on the number of samples played out with the default settings, and a lot more when URBs convey more packets to reduce the interrupt rate and power consumption. This patch relies on the USB frame counter to reduce the uncertainty to less than 2ms worst-case. The delay information essentially becomes independent of the URB size and number of packets. This should help applications like PulseAudio which require accurate audio timing. Clemens Ladisch reported a decrease of mplayer's A-V difference from nrpacks down to at most 1ms. Thanks to Clemens for also pointing out that the implementation of frame counters varies between different HCDs. Only the 8 lowest-bits are used to estimate the delay. Signed-off-by: NPierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> [clemens: changed debug code] Signed-off-by: NClemens Ladisch <clemens@ladisch.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 11 3月, 2011 1 次提交
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由 Oliver Neukum 提交于
Devices are autosuspended if no pcm nor midi channel is open Mixer devices may be opened. This way they are active when in use to play or record sound, but can be suspended while users have a mixer application running. [Small clean-ups using static inline by tiwai] Signed-off-by: NOliver Neukum <oneukum@suse.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 23 2月, 2011 1 次提交
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由 Takashi Iwai 提交于
When a USB audio device is disconnected, snd_usb_audio_disconnect() kills all audio URBs. At the same time, the application, after being notified of the disconnection, might close the device, in which case ALSA calls the .hw_free callback, which should free the URBs too. Commit de1b8b93 "[ALSA] Fix hang-up at disconnection of usb-audio" prevented snd_usb_hw_free() from freeing the URBs to avoid a hang that resulted from this race, but this introduced another race because the URB callbacks could now be executed after snd_usb_hw_free() has returned, and try to access already freed data. Fix the first race by introducing a mutex to serialize the disconnect callback and all PCM callbacks that manage URBs (hw_free and hw_params). Reported-and-tested-by: NPierre-Louis Bossart <pierre-louis.bossart@intel.com> Cc: <stable@kernel.org> [CL: also serialize hw_params callback] Signed-off-by: NClemens Ladisch <clemens@ladisch.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 01 11月, 2010 1 次提交
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由 Jesper Juhl 提交于
sound/usb/pcm.c::snd_usb_pcm_check_knot() fails to check the return value from kmalloc() and may end up dereferencing a null pointer. The patch below (compile tested only) should take care of that little problem. Signed-off-by: NJesper Juhl <jj@chaosbits.net> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 27 10月, 2010 1 次提交
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由 Clemens Ladisch 提交于
There are two USB Audio Class specifications (v1 and v2), but neither of them clearly defines the feedback format for high-speed UAC v1 devices. Add to this whatever the Creative and M-Audio firmware writers have been smoking, and it becomes impossible to predict the exact feedback format used by a particular device. Therefore, automatically detect the feedback format by looking at the magnitude of the first received feedback value. Also, this allows us to get rid of some special cases for E-Mu devices. Signed-off-by: NClemens Ladisch <clemens@ladisch.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 04 9月, 2010 1 次提交
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由 Clemens Ladisch 提交于
The Audio Class v2 support code in 2.6.35 added checks for the bInterfaceProtocol field. However, there are devices (usually those detected by vendor-specific quirks) that do not have one of the predefined values in this field, which made the driver reject them. To fix this regression, restore the old behaviour, i.e., assume that a device with an unknown bInterfaceProtocol field (other than UAC_VERSION_2) has more or less UAC-v1-compatible descriptors. [compile warning fixes by tiwai] Signed-off-by: NClemens Ladisch <clemens@ladisch.de> Cc: Daniel Mack <daniel@caiaq.de> Cc: <stable@kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 14 8月, 2010 1 次提交
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由 Paul Zimmerman 提交于
This is V2 of the patch, after feedback from Clemens and Daniel. This patch adds SuperSpeed support to the USB drivers under sound/. It adds tests for USB_SPEED_SUPER to the appropriate places that check for the USB speed. This patch has been tested with our SS USB3 device emulating a set of Yamaha speakers and a Logitech microphone, but with the descriptors modified to add USB3 support. It has also been tested with the real speakers and microphone, to make sure that USB2 devices still work. Signed-off-by: NPaul Zimmerman <paulz@synopsys.com> Cc: Clemens Ladisch <clemens@ladisch.de> Cc: Daniel Mack <daniel@caiaq.de> Cc: Greg Kroah-Hartman <gregkh@suse.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 13 7月, 2010 1 次提交
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由 Uwe Kleine-König 提交于
Signed-off-by: NUwe Kleine-König <u.kleine-koenig@pengutronix.de> Signed-off-by: NJiri Kosina <jkosina@suse.cz>
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- 01 6月, 2010 1 次提交
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由 Daniel Mack 提交于
Audio devices which comply to the UAC2 standard can export complex clock topologies in its descriptors and set up links between them. The entities that are defined are - clock sources, which define the end-leafs. - clock selectors, which act as switch to select one out of many possible clocks sources. - clock multipliers, which have an input clock source, and act as clock source again. They can be used to derive one clock from another. All sample rate changes, clock validity queries and the like must go to clock source elements, while clock selectors and multipliers can be used as terminal clock source. The following patch adds a parser for these elements and functions to iterate over the tree and find the leaf nodes (clock sources). The samplerate set functions were moved to the new clock.c file. Signed-off-by: NDaniel Mack <daniel@caiaq.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 27 5月, 2010 1 次提交
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由 Daniel Mack 提交于
This request is again handled differently in comparison to UAC1. Signed-off-by: NDaniel Mack <daniel@caiaq.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 29 3月, 2010 1 次提交
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由 Stephen Rothwell 提交于
Signed-off-by: NStephen Rothwell <sfr@canb.auug.org.au> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 12 3月, 2010 1 次提交
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由 Daniel Mack 提交于
- Split the audio.h file in two to clearly denote the differences between the standards. - Add many more defines to audio-v2.h. Most of them are not currently used. - Replaced a magic value with a proper define Signed-off-by: NDaniel Mack <daniel@caiaq.de> Acked-by: NGreg Kroah-Hartman <gregkh@suse.de> Cc: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 05 3月, 2010 4 次提交
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由 Daniel Mack 提交于
Sample rate setting is done with a 4-byte long class request that addresses the interface. Signed-off-by: NDaniel Mack <daniel@caiaq.de> Cc: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Clemens Ladisch 提交于
In preparation for USB audio 2.0 support, change the audioformat structure so that it uses a bitmask to specify possible formats. Signed-off-by: NClemens Ladisch <clemens@ladisch.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Clemens Ladisch 提交于
The snd_usb_substream::format field actually contains the index of the current alternate setting, so rename it to altset_idx to avoid confusion. Signed-off-by: NClemens Ladisch <clemens@ladisch.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Daniel Mack 提交于
Clean up the usb audio driver by factoring out a lot of functions to separate files. Code for procfs, quirks, urbs, format parsers etc all got a new home now. Moved almost all special quirk handling to quirks.c and introduced new generic functions to handle them, so the exceptions do not pollute the whole driver. Renamed usbaudio.c to card.c because this is what it actually does now. Renamed usbmidi.c to midi.c for namespace clarity. Removed more things from usbaudio.h. The non-standard drivers were adopted accordingly. Signed-off-by: NDaniel Mack <daniel@caiaq.de> Cc: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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