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- 06 Sep, 2012 1 commit
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Takashi Iwai authored
The recent fix for the missing fine delayed time adjustment gives strange error messages at each start of the playback stream, such as delay: estimated 0, actual 352 delay: estimated 353, actual 705 These come from the sanity check in retire_playback_urb(). Before the stream is activated via start_endpoints(), a few silent packets have been already sent. And at this point the delay account is still in the state as if the new packets are just queued, so the driver gets confused and spews the bogus error messages. For fixing the issue, we just need to check whether the received packet is valid, whether it's zero sized or not. Reported-by:
Markus Trippelsdorf <markus@trippelsdorf.de> Cc: <stable@vger.kernel.org> [v3.5+] Signed-off-by:
Takashi Iwai <tiwai@suse.de>
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- 31 Aug, 2012 3 commits
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Daniel Mack authored
Commit 68e67f40 ("ALSA: snd-usb: move calls to usb_set_interface") saved us some unnecessary calls to snd_usb_set_interface() but ignored the fact that there is at least one device out there which operates on two endpoint in different interfaces simultaniously. Take care for this by catching the case where data and sync endpoints are located on different interfaces and calling snd_usb_set_interface() between the start of the two endpoints. Signed-off-by:
Daniel Mack <zonque@gmail.com> Reported-by:
Robert M. Albrecht <linux@romal.de> Cc: stable@kernel.org [v3.5+] Signed-off-by:
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Daniel Mack authored
In order to support devices with implicit feedback streaming models, packet sizes are now stored with each individual urb, and the PCM handling code which fills the buffers purely relies on the size fields now. However, calling snd_usb_audio_next_packet_size() for all possible packets in an URB at once, prior to letting the PCM code do its job does in fact not lead to the same behaviour than what the old code did: The PCM code will break its loop once a period boundary is reached, consequently using up less packets that it really could. As snd_usb_audio_next_packet_size() implements a feedback mechanism to the endpoints phase accumulator, the number of calls to that function matters, and when called too often, the data rate runs out of bounds. Fix this by making the next_packet function public, and call it from the PCM code as before if the packet data sizes are not defined. Signed-off-by:
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Daniel Mack authored
Parts of commit 294c4fb8 ("ALSA: usb: refine delay information with USB frame counter") were unfortunately lost during the refactoring of the snd-usb driver in 3.5. This patch adds them back, restoring the correct delay information behaviour. Signed-off-by:
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- 30 Aug, 2012 1 commit
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Daniel Mack authored
Commit e9ba389c ("ALSA: usb-audio: Fix scheduling-while-atomic bug in PCM capture stream") fixed a scheduling-while-atomic bug that happened when snd_usb_endpoint_start was called from the trigger callback, which is an atmic context. However, the patch breaks the idea of the endpoints reference counting, which is the reason why the driver has been refactored lately. Revert that commit and let snd_usb_endpoint_start() take care of the URB cancellation again. As this function is called from both atomic and non-atomic context, add a flag to denote whether the function may sleep. Signed-off-by:
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- 16 Aug, 2012 1 commit
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Takashi Iwai authored
A PCM capture stream on usb-audio causes a scheduling-while-atomic BUG, as reported in the bugzilla entry below. It's because snd_usb_endpoint_start() is called at first at trigger START for a capture stream, and this function contains the left-over EP deactivation codes. The problem doesn't happen for a playback stream because the function is called at PCM prepare time, which can sleep. This patch fixes the BUG by moving the EP deactivation code into the PCM prepare callback. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=46011 Cc: <stable@vger.kernel.org> [v3.5+] Signed-off-by:
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- 13 Jul, 2012 1 commit
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Daniel Mack authored
The rework of the snd-usb endpoint logic moved the calls to snd_usb_set_interface() into the snd_usb_endpoint implemenation. This changed the order in which these calls are issued to the device, and thereby caused regressions for some webcams. Fix this by moving the calls back to pcm.c for now to make it work again and use snd_usb_endpoint_activate() to really tear down all remaining URBs in the flight, consequently fixing another regression caused by USB packets on the wire after altsetting 0 has been selected. Signed-off-by:
Philipp Dreimann <philipp@dreimann.net> Reported-by:
Joseph Salisbury <joseph.salisbury@canonical.com> Signed-off-by:
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- 06 Jul, 2012 1 commit
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Takashi Iwai authored
In the new PCM streaming logic, the interface number is assigned to usb stream instance (subs->interface) after the format and rate setups are succeeded, but some codes are still passing subs->interface as the reference to helper functions. This leads to initializing with an invalid iface number (-1). This patch replaces the wrong references with the ones from the target fmt correctly. Signed-off-by:
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- 18 Jun, 2012 2 commits
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Daniel Mack authored
Signed-off-by:
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Daniel Mack authored
Fix a bogus sanity check for sync pipe in pcm.c. This flaw was introduced during the streaming logic refactorization. While at it, improve the error messages that are generated in such cases. Signed-off-by:
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- 31 May, 2012 1 commit
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Clemens Ladisch authored
The array of sample rates is reallocated every time when opening the PCM device, but was freed only once when unplugging the device. Reported-by:
"Alexander E. Patrakov" <patrakov@gmail.com> Cc: <stable@vger.kernel.org> Signed-off-by:
Clemens Ladisch <clemens@ladisch.de> Signed-off-by:
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- 21 May, 2012 1 commit
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Daniel Mack authored
Set some substream struct members to make the proc interface code work again. Signed-off-by:
Felix Homann <linuxaudio@showlabor.de> Signed-off-by:
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- 13 Apr, 2012 2 commits
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Daniel Mack authored
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:
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Daniel Mack authored
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:
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- 15 Mar, 2012 1 commit
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Takashi Iwai authored
Signed-off-by:
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- 27 Sep, 2011 1 commit
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Clemens Ladisch authored
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:
Felipe Balbi <balbi@ti.com> Signed-off-by:
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- 14 Sep, 2011 1 commit
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Daniel Mack authored
No code altered at this point, simply preparing for upcoming refactorizations. Signed-off-by:
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- 12 Sep, 2011 1 commit
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Pierre-Louis Bossart authored
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:
Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> [clemens: changed debug code] Signed-off-by:
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- 11 Mar, 2011 1 commit
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Oliver Neukum authored
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:
Oliver Neukum <oneukum@suse.de> Signed-off-by:
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- 23 Feb, 2011 1 commit
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Takashi Iwai authored
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:
Pierre-Louis Bossart <pierre-louis.bossart@intel.com> Cc: <stable@kernel.org> [CL: also serialize hw_params callback] Signed-off-by:
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- 01 Nov, 2010 1 commit
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Jesper Juhl authored
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:
Jesper Juhl <jj@chaosbits.net> Signed-off-by:
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- 27 Oct, 2010 1 commit
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Clemens Ladisch authored
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:
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- 03 Sep, 2010 1 commit
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Clemens Ladisch authored
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:
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- 14 Aug, 2010 1 commit
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Paul Zimmerman authored
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:
Paul 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:
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- 12 Jul, 2010 1 commit
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Uwe Kleine-König authored
Signed-off-by:
Uwe Kleine-König <u.kleine-koenig@pengutronix.de> Signed-off-by:
Jiri Kosina <jkosina@suse.cz>
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- 31 May, 2010 1 commit
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Daniel Mack authored
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:
Daniel Mack <daniel@caiaq.de> Signed-off-by:
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- 27 May, 2010 1 commit
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Daniel Mack authored
This request is again handled differently in comparison to UAC1. Signed-off-by:
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- 29 Mar, 2010 1 commit
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Stephen Rothwell authored
Signed-off-by:
Stephen Rothwell <sfr@canb.auug.org.au> Signed-off-by:
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- 12 Mar, 2010 1 commit
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Daniel Mack authored
- 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:
Greg Kroah-Hartman <gregkh@suse.de> Cc: Clemens Ladisch <clemens@ladisch.de> Signed-off-by:
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- 05 Mar, 2010 4 commits
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Daniel Mack authored
Sample rate setting is done with a 4-byte long class request that addresses the interface. Signed-off-by:
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Clemens Ladisch authored
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:
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Clemens Ladisch authored
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:
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Daniel Mack authored
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:
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