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* ALSA: usb-audio: Properly refcounting clock rateTakashi Iwai2022-10-211-4/+7
| | | | | | | | | | | | | | | | | | | [ Upstream commit 9a737e7f8b371e97eb649904276407cee2c9cf30 ] We fixed the bug introduced by the patch for managing the shared clocks at the commit 809f44a0cc5a ("ALSA: usb-audio: Clear fixed clock rate at closing EP"), but it was merely a workaround. By this change, the clock reference rate is cleared at each EP close, hence the still remaining EP may need a re-setup of rate unnecessarily. This patch introduces the proper refcounting for the clock reference object so that the clock setup is done only when needed. Fixes: 809f44a0cc5a ("ALSA: usb-audio: Clear fixed clock rate at closing EP") Fixes: c11117b634f4 ("ALSA: usb-audio: Refcount multiple accesses on the single clock") Link: https://lore.kernel.org/r/20220920181126.4912-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ALSA: usb-audio: Fix NULL dererence at error pathTakashi Iwai2022-10-211-1/+2
| | | | | | | | | | | | | | | | | | commit 568be8aaf8a535f79c4db76cabe17b035aa2584d upstream. At an error path to release URB buffers and contexts, the driver might hit a NULL dererence for u->urb pointer, when u->buffer_size has been already set but the actual URB allocation failed. Fix it by adding the NULL check of urb. Also, make sure that buffer_size is cleared after the error path or the close. Cc: <stable@vger.kernel.org> Reported-by: Sabri N. Ferreiro <snferreiro1@gmail.com> Link: https://lore.kernel.org/r/CAKG+3NRjTey+fFfUEGwuxL-pi_=T4cUskYG9OzpzHytF+tzYng@mail.gmail.com Link: https://lore.kernel.org/r/20220930100129.19445-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: usb-audio: Fix potential memory leaksTakashi Iwai2022-10-211-2/+1
| | | | | | | | | | | | | | | | | | commit 6382da0828995af87aa8b8bef28cc61aceb4aff3 upstream. When the driver hits -ENOMEM at allocating a URB or a buffer, it aborts and goes to the error path that releases the all previously allocated resources. However, when -ENOMEM hits at the middle of the sync EP URB allocation loop, the partially allocated URBs might be left without released, because ep->nurbs is still zero at that point. Fix it by setting ep->nurbs at first, so that the error handler loops over the full URB list. Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20220930100151.19461-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* Revert "ALSA: usb-audio: Split endpoint setups for hw_params and prepare"Takashi Iwai2022-09-201-9/+14
| | | | | | | | | | | | | | | | This reverts commit ff878b408a03bef5d610b7e2302702e16a53636e. Unfortunately the recent fix seems bringing another regressions with PulseAudio / pipewire, at least for Steinberg and MOTU devices. As a temporary solution, do a straight revert. The issue for Android will be revisited again later by another different fix (if any). Fixes: ff878b408a03 ("ALSA: usb-audio: Split endpoint setups for hw_params and prepare") Cc: <stable@vger.kernel.org> BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=216500 Link: https://lore.kernel.org/r/20220920113929.25162-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Clear fixed clock rate at closing EPTakashi Iwai2022-09-071-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | The recent commit c11117b634f4 ("ALSA: usb-audio: Refcount multiple accesses on the single clock") tries to manage the clock rate shared by several endpoints. This was intended for avoiding the unmatched rate by a different endpoint, but unfortunately, it introduced a regression for PulseAudio and pipewire, too; those applications try to probe the multiple possible rates (44.1k and 48kHz) and setting up the normal rate fails but only the last rate is applied. The cause is that the last sample rate is still left to the clock reference even after closing the endpoint, and this value is still used at the next open. It happens only when applications set up via PCM prepare but don't start/stop the stream; the rate is reset when the stream is stopped, but it's not cleared at close. This patch addresses the issue above, simply by clearing the rate set in the clock reference at the last close of each endpoint. Fixes: c11117b634f4 ("ALSA: usb-audio: Refcount multiple accesses on the single clock") Reported-by: Jason A. Donenfeld <Jason@zx2c4.com> Tested-by: Jason A. Donenfeld <Jason@zx2c4.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/all/YxXIWv8dYmg1tnXP@zx2c4.com/ Link: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/2620 Link: https://lore.kernel.org/r/20220907100421.6443-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Split endpoint setups for hw_params and prepareTakashi Iwai2022-09-011-14/+9
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | One of the former changes for the endpoint management was the more consistent setup of endpoints at hw_params. snd_usb_endpoint_configure() is a single function that does the full setup, and it's called from both PCM hw_params and prepare callbacks. Although the EP setup at the prepare phase is usually skipped (by checking need_setup flag), it may be still effective in some cases like suspend/resume that requires the interface setup again. As it's a full and single setup, the invocation of snd_usb_endpoint_configure() includes not only the USB interface setup but also the buffer release and allocation. OTOH, doing the buffer release and re-allocation at PCM prepare phase is rather superfluous, and better to be done only in the hw_params phase. For those optimizations, this patch splits the endpoint setup to two phases: snd_usb_endpoint_set_params() and snd_usb_endpoint_prepare(), to be called from hw_params and from prepare, respectively. Note that this patch changes the driver operation slightly, effectively moving the USB interface setup again to PCM prepare stage instead of hw_params stage, while the buffer allocation and such initializations are still done at hw_params stage. And, the change of the USB interface setup timing (moving to prepare) gave an interesting "fix", too: it was reported that the recent kernels caused silent output at the beginning on playbacks on some devices on Android, and this change casually fixed the regression. It seems that those devices are picky about the sample rate change (or the interface change?), and don't follow the too immediate rate changes. Meanwhile, Android operates the PCM in the following order: - open, then hw_params with the possibly highest sample rate - close without prepare - re-open, hw_params with the normal sample rate - prepare, and start streaming This procedure ended up the hw_params twice with different rates, and because the recent kernel did set up the sample rate twice one and after, it screwed up the device. OTOH, the earlier kernels didn't set up the USB interface at hw_params, hence this problem didn't appear. Now, with this patch, the USB interface setup is again back to the prepare phase, and it works around the problem automagically. Although we should address the sample rate problem in a more solid way in future, let's keep things working as before for now. Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management") Cc: <stable@vger.kernel.org> Reported-by: chihhao chen <chihhao.chen@mediatek.com> Link: https://lore.kernel.org/r/87e6d6ae69d68dc588ac9acc8c0f24d6188375c3.camel@mediatek.com Link: https://lore.kernel.org/r/20220901124136.4984-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Use atomic_try_cmpxchg in ep_state_updateUros Bizjak2022-07-141-1/+1
| | | | | | | | | | | | | Use atomic_try_cmpxchg instead of atomic_cmpxchg (*ptr, old, new) == old in ep_state_update. x86 CMPXCHG instruction returns success in ZF flag, so this change saves a compare after cmpxchg (and related move instruction in front of cmpxchg). No functional change intended. Signed-off-by: Uros Bizjak <ubizjak@gmail.com> Link: https://lore.kernel.org/r/20220713151946.4743-1-ubizjak@gmail.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Fix wrong kfree issue in snd_usb_endpoint_free_allWan Jiabing2022-05-181-1/+1
| | | | | | | | | | | | Fix following coccicheck error: ./sound/usb/endpoint.c:1671:8-10: ERROR: reference preceded by free on line 1671. Here should be 'cp' rather than 'ip'. Fixes: c11117b634f4 ("ALSA: usb-audio: Refcount multiple accesses on the single clock") Signed-off-by: Wan Jiabing <wanjiabing@vivo.com> Link: https://lore.kernel.org/r/20220518021617.10114-1-wanjiabing@vivo.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Refcount multiple accesses on the single clockTakashi Iwai2022-05-161-9/+81
| | | | | | | | | | | | | | | | | When a clock source is connected to multiple nodes / endpoints, the current USB-audio driver tries to set up at each time one of them is configured. Although it reads the current rate and updates only if it differs, some devices seem unhappy with this behavior and spew the errors when reading/updating the rate unnecessarily. This patch tries to reduce the redundant clock setup by introducing a refcount for each clock source. When the stream is actually running, a clock rate is "locked", and it bypasses the clock and/or refuse to change any longer. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=215934 Link: https://lore.kernel.org/r/20220516104807.16482-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Fix packet size calculation regressionTakashi Iwai2021-10-011-1/+1
| | | | | | | | | | | | | The commit d215f63d49da ("ALSA: usb-audio: Check available frames for the next packet size") introduced the available frame size check, but the conversion forgot to initialize the temporary variable properly, and it resulted in a bogus calculation. This patch fixes it. Fixes: d215f63d49da ("ALSA: usb-audio: Check available frames for the next packet size") Reported-by: Colin Ian King <colin.king@canonical.com> Link: https://lore.kernel.org/r/20211001104417.14291-1-colin.king@canonical.com Link: https://lore.kernel.org/r/20211001105425.16191-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Avoid killing in-flight URBs during drainingTakashi Iwai2021-09-301-5/+9
| | | | | | | | | | | | | | | | | | | While draining a stream, ALSA PCM core stops the stream by issuing snd_pcm_stop() after all data has been sent out. And, at PCM trigger stop, currently USB-audio driver kills the in-flight URBs explicitly, then at sync-stop ops, sync with the finish of all remaining URBs. This might result in a drop of the drained samples as most of USB-audio devices / hosts allow relatively long in-flight samples (as a sort of FIFO). For avoiding the trimming, this patch changes the stream-stop behavior during PCM draining state. Under that condition, the pending URBs won't be killed. The leftover in-flight URBs are caught by the sync-stop operation that shall be performed after the trigger-stop operation. Link: https://lore.kernel.org/r/20210929080844.11583-10-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Improved lowlatency playback supportTakashi Iwai2021-09-301-42/+88
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This is another attempt to improve further the handling of playback stream in the low latency mode. The latest workaround in commit 4267c5a8f313 ("ALSA: usb-audio: Work around for XRUN with low latency playback") revealed that submitting URBs forcibly in advance may trigger XRUN easily. In the classical mode, this problem was avoided by practically delaying the submission of the actual data with the pre-submissions of silent data before triggering the stream start. But that is exactly what we want to avoid. Now, in this patch, instead of the previous workaround, we take a similar approach as used in the implicit feedback mode. The URBs are queued at the PCM trigger start like before, but we check whether the buffer has been already filled enough before each submission, and stop queuing if the data overcomes the threshold. The remaining URBs are kept in the ready list, and they will be retrieved in the URB complete callback of other (already queued) URBs. In the complete callback, we try to fill the data and submit as much as possible again. When there is no more available in-flight URBs that may handle the pending data, we'll check in PCM ack callback and submit and process URBs there in addition. In this way, the amount of in-flight URBs may vary dynamically and flexibly depending on the available data without hitting XRUN. The following things are changed to achieve the behavior above: * The endpoint prepare callback is changed to return an error code; when there is no enough data available, it may return -EAGAIN. Currently only prepare_playback_urb() returns the error. The evaluation of the available data is a bit messy here; we can't check with snd_pcm_avail() at the point of prepare callback (as runtime->status->hwptr hasn't been updated yet), hence we manually estimate the appl_ptr and compare with the internal hwptr_done to calculate the available frames. * snd_usb_endpoint_start() doesn't submit full URBs if the prepare callback returns -EAGAIN, and puts the remaining URBs to the ready list for the later submission. * snd_complete_urb() treats the URBs in the low-latency mode similarly like the implicit feedback mode, and submissions are done in (now exported) snd_usb_queue_pending_output_urbs(). * snd_usb_queue_pending_output_urbs() again checks the error value from the prepare callback. If it's -EAGAIN for the normal stream (i.e. not implicit feedback mode), we push it back to the ready list again. * PCM ack callback is introduced for the playback stream, and it calls snd_usb_queue_pending_output_urbs() if there is no in-flight URB while the stream is running. This corresponds to the case where the system needs the appl_ptr update for re-submitting a new URB. * snd_usb_queue_pending_output_urbs() and the prepare EP callback receive in_stream_lock argument, which is a bool flag indicating the call path from PCM ack. It's needed for avoiding the deadlock of snd_pcm_period_elapsed() calls. * Set the new SNDRV_PCM_INFO_EXPLICIT_SYNC flag when the new low-latency mode is deployed. This assures catching each applptr update even in the mmap mode. Fixes: 4267c5a8f313 ("ALSA: usb-audio: Work around for XRUN with low latency playback") Link: https://lore.kernel.org/r/20210929080844.11583-9-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Add spinlock to stop_urbs()Takashi Iwai2021-09-301-0/+3
| | | | | | | | | In theory, stop_urbs() may be called concurrently. Although we have the state check beforehand, it's safer to apply ep->lock during the critical list head manipulations. Link: https://lore.kernel.org/r/20210929080844.11583-8-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Check available frames for the next packet sizeTakashi Iwai2021-09-301-15/+36
| | | | | | | | | | | | | | | This is yet more preparation for the upcoming changes. Extend snd_usb_endpoint_next_packet_size() to check the available frames and return -EAGAIN if the next packet size is equal or exceeds the given size. This will be needed for avoiding XRUN during the low latency operation. As of this patch, avail=0 is passed, i.e. the check is skipped and no behavior change. Link: https://lore.kernel.org/r/20210929080844.11583-7-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Rename early_playback_start flag with lowlatency_playbackTakashi Iwai2021-09-301-0/+4
| | | | | | | | | | | | | | | | | | This is a preparation patch for the upcoming low-latency improvement changes. Rename early_playback_start flag with lowlatency_playback as it's more intuitive. The new flag is basically a reverse meaning. Along with the rename, factor out the code to set the flag to a function. This makes the complex condition checks simpler. Also, the same flag is introduced to snd_usb_endpoint, too, that is carried from the snd_usb_substream flag. Currently the endpoint flag isn't still referred, but will be used in later patches. Link: https://lore.kernel.org/r/20210929080844.11583-4-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Fix possible race at sync of urb completionsTakashi Iwai2021-09-301-1/+6
| | | | | | | | | | | | | | | | | | | | | USB-audio driver tries to sync with the clear of all pending URBs in wait_clear_urbs(), and it waits for all bits in active_mask getting cleared. This works fine for the normal operations, but when a stream is managed in the implicit feedback mode, there is still a very thin race window: namely, in snd_complete_usb(), the active_mask bit for the current URB is once cleared before re-submitted in queue_pending_output_urbs(). If wait_clear_urbs() is called during that period, it may pass the test and go forward even though there may be a still pending URB. For covering it, this patch adds a new counter to each endpoint to keep the number of in-flight URBs, and changes wait_clear_urbs() checking this number instead. The counter is decremented at the end of URB complete, hence the reference is kept as long as the URB complete is in process. Link: https://lore.kernel.org/r/20210929080844.11583-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Restrict rates for the shared clocksTakashi Iwai2021-09-301-0/+21
| | | | | | | | | | | | | When a single clock source is shared among several endpoints, we have to keep the same rate on all active endpoints as long as the clock is being used. For dealing with such a case, this patch adds one more check in the hw params constraint for the rate to take the shared clocks into account. The current rate is evaluated from the endpoint list that applies the same clock source. BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1190418 Link: https://lore.kernel.org/r/20210929080844.11583-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Merge branch 'for-linus' into for-nextTakashi Iwai2021-08-301-0/+4
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| * ALSA: usb-audio: Work around for XRUN with low latency playbackTakashi Iwai2021-08-271-0/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The recent change for low latency playback works in most of test cases but it turned out still to hit errors on some use cases, most notably with JACK with small buffer sizes. This is because USB-audio driver fills up and submits full URBs at the beginning, while the URBs would return immediately and try to fill more -- that can easily trigger XRUN. It was more or less expected, but in the small buffer size, the problem became pretty obvious. Fixing this behavior properly would require the change of the fundamental driver design, so it's no trivial task, unfortunately. Instead, here we work around the problem just by switching back to the old method when the given configuration is too fragile with the low latency stream handling. As a threshold, we calculate the total buffer bytes in all plus one URBs, and check whether it's beyond the PCM buffer bytes. The one extra URB is needed because XRUN happens at the next submission after the first round. Fixes: 307cc9baac5c ("ALSA: usb-audio: Reduce latency at playback start, take#2") Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20210827203311.5987-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: usb-audio: Move set-interface-first workaround into common quirkTakashi Iwai2021-08-241-4/+2
| | | | | | | | | | | | | | | | | | | | | | The recent quirk for WALKMAN (commit 7af5a14371c1: "ALSA: usb-audio: Fix regression on Sony WALKMAN NW-A45 DAC") may be required for other devices and is worth to be put into the common quirk flags. This patch adds a new quirk flag bit QUIRK_FLAG_SET_IFACE_FIRST and a quirk table entry for the device. Link: https://lore.kernel.org/r/20210824055720.9240-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | Merge branch 'for-linus' into for-nextTakashi Iwai2021-08-241-0/+5
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| * ALSA: usb-audio: Fix regression on Sony WALKMAN NW-A45 DACTakashi Iwai2021-08-241-0/+5
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | We've got a regression report for USB-audio with Sony WALKMAN NW-A45 DAC device where no sound is audible on recent kernel. The bisection resulted in the code change wrt endpoint management, and the further debug session revealed that it was caused by the order of the USB audio interface. In the earlier code, we always set up the USB interface at first before other setups, but it was changed to be done at the last for UAC2/3, which is more standard way, while keeping the old way for UAC1. OTOH, this device seems requiring the setup of the interface at first just like UAC1. This patch works around the regression by applying the interface setup specifically for the WALKMAN at the beginning of the endpoint setup procedure. This change is written straightforwardly to be easily backported in old kernels. A further cleanup to move the workaround into a generic quirk section will follow in a later patch. Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management") Cc: <stable@vger.kernel.org> BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=214105 Link: https://lore.kernel.org/r/20210824054700.8236-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: usb-audio: Move interface setup delay into quirk_flagsTakashi Iwai2021-08-021-1/+2
| | | | | | | | | | | | | | | | Yet another delay is applied at switching the interface. This can be moved to quirk_flags, too. Link: https://lore.kernel.org/r/20210729073855.19043-10-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: usb-audio: Move playback_first flag into quirk_flagsTakashi Iwai2021-08-021-1/+1
| | | | | | | | | | | | | | | | The snd_usb_audio.playback_first flag is used by the implicit feedback mode handling, and this can be also moved to quirk_flags. Link: https://lore.kernel.org/r/20210729073855.19043-6-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: usb-audio: Move tx_length quirk handling to quirk_flagsTakashi Iwai2021-08-021-2/+7
|/ | | | | | | | | There is another quirk for the transfer, and that's currently specific to Zoom R16/24, handled in create_standard_audio_quirk(). Let's move this also to the new quirk_flags. Link: https://lore.kernel.org/r/20210729073855.19043-5-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: fix spelling mistakesgushengxian2021-07-051-2/+2
| | | | | | | | | | | | | | Fix some spelling mistakes as follows: altenate ==> alternate compatbile ==> compatible perfoms ==> performs dont'register ==> don't register periodicaly ==> periodically arount ==> around Signed-off-by: gushengxian <gushengxian@yulong.com> Link: https://lore.kernel.org/r/20210705120052.665212-1-gushengxian507419@gmail.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Refactoring delay account codeTakashi Iwai2021-06-021-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The PCM delay accounting in USB-audio driver is a bit complex to follow, and this is an attempt to improve the readability and provide some potential fix. Basically, the PCM position delay is calculated from two factors: the in-flight data on URBs and the USB frame counter. For the playback stream, we advance the hwptr already at submitting URBs. Those "in-flight" data amount is now tracked, and this is used as the base value for the PCM delay correction. The in-flight data is decreased again at URB completion in return. For the capture stream, OTOH, there is no in-flight data, hence the delay base is zero. The USB frame counter is used in addition for correcting the current position. The reference frame counter is updated at each submission and receiving time, and the difference from the current counter value is taken into account. In this patch, each in-flight data bytes is recorded in the new snd_usb_ctx.queued field, and the total in-flight amount is tracked in snd_usb_substream.inflight_bytes field, as the replacement of last_delay field. Note that updating the hwptr after URB completion doesn't work for PulseAudio who tries to scratch the buffer on the fly; USB-audio is basically a double-buffer implementation, hence the scratching the buffer can't work for the already submitted data. So we always update hwptr beforehand. It's not ideal, but the delay account should give enough correctness. Link: https://lore.kernel.org/r/20210601162457.4877-4-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Fix implicit sync clearance at stopping streamTakashi Iwai2021-04-261-4/+4
| | | | | | | | | | | | | | | | | | | The recent endpoint management change for implicit feedback mode added a clearance of ep->sync_sink (formerly ep->sync_slave) pointer at snd_usb_endpoint_stop() to assure no leftover for the feedback from the already stopped capture stream. This turned out to cause a regression, however, when full-duplex streams were running and only a capture was stopped. Because of the above clearance of ep->sync_sink pointer, no more feedback is done, hence the playback will stall. This patch fixes the ep->sync_sink clearance to be done only after all endpoints are released, for addressing the regression. Reported-and-tested-by: Lucas Endres <jaffa225man@gmail.com> Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management") Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20210426063349.18601-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Apply implicit feedback mode for BOSS devicesTakashi Iwai2021-04-141-1/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | During the recent rewrite of the implicit feedback support, we've tested to apply the implicit fb on BOSS devices, but it failed, as the capture stream didn't start without the playback. As the end result, it got another type of quirk for tying both streams but starts playback always (commit 6234fdc1cede "ALSA: usb-audio: Quirk for BOSS GT-001"). Meanwhile, Mike Oliphant has tested the real implicit feedback mode for the playback again with the latest code, and found out that it actually works if the initial feedback sync is skipped; that is, on those BOSS devices, the playback stream has to be started at first without waiting for the capture URB completions. Otherwise it gets stuck. In the rest operations after the capture stream processed, we can take them as the implicit feedback source. This patch is an attempt to improve the support for BOSS devices with the implicit feedback mode in the way described above. It adds a new flag to snd_usb_audio, playback_first, indicating that the playback stream starts without sync with the initial capture completion. This flag is set in the quirk table with the new IMPLICIT_FB_BOTH type. Reported-and-tested-by: Mike Oliphant <oliphant@nostatic.org> Link: https://lore.kernel.org/r/20210414083255.9527-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Don't avoid stopping the stream at disconnectionTakashi Iwai2021-02-081-3/+0
| | | | | | | | | | | | | In the later patch, we're going to issue the PCM sync_stop calls at disconnection. But currently the USB-audio driver can't handle it because it has a check of shutdown flag for stopping the URBs. This is basically superfluous (the stopping URBs are safe at disconnection state), so let's drop the check. Fixes: dc5eafe7787c ("ALSA: usb-audio: Support PCM sync_stop") Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20210206203052.15606-4-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: More strict state change in EPTakashi Iwai2021-02-081-14/+28
| | | | | | | | | | | | | | | | | | | The endpoint management has bit flags to indicate the current state, and we're dealing two things: the running bit and the stopping bit. There is a thin window in transition from the running to the stopping in stop_urbs(), and as long as the bit flags are used, it's difficult to plug. This patch modifies the state management code to use the atomic int and follow the explicit three states, STOPPED, RUNNING and STOPPING. The state change is done via atomic_cmpxhg() for avoiding possible races, and check the state change more strictly. The unexpected state change is now handled as an error. Fixes: d0f09d1e4a88 ("ALSA: usb-audio: Refactoring endpoint URB deactivation") Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20210206203052.15606-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Handle invalid running state at releasing EPTakashi Iwai2021-02-081-19/+21
| | | | | | | | | | | | | | | | | | | When we stop an endpoint in release_urbs(), it ignores the inconsistent endpoint state and tries to release the resources. This shouldn't happen in theory, but it's still safer to abort the release and let the caller proper error handling. Also, stop_and_unlink_urbs() called from release_urbs() does two step works, and it's more straightforward to split this to two functions again, so that the call from the PCM trigger won't take the path with sleeping. This patch modifies the EP management code to adapt two points above. Fixes: d0f09d1e4a88 ("ALSA: usb-audio: Refactoring endpoint URB deactivation") Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20210206203052.15606-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Correct document for snd_usb_endpoint_free_all()Takashi Iwai2021-02-051-1/+1
| | | | | | | | | | | The kerndoc comment for the new function snd_usb_endpoint_free_all() had a typo wrt the argument name. Fix it. Fixes: 00272c61827e ("ALSA: usb-audio: Avoid unnecessary interface re-setup") Reported-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20210205082837.6327-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Set sample rate for all sharing EPs on UAC1Takashi Iwai2021-01-181-0/+9
| | | | | | | | | | | | | | | | The UAC2/3 sample rate setup is based on the clock node, which is usually shared in the interface, and can't be re-setup without deselecting the interface once, and that's how the current code behaves. OTOH, the sample rate setup of UAC1 is per endpoint, hence we basically need to call for each endpoint usage even if those share the same interface. This patch fixes the behavior of UAC1 to call always snd_usb_init_sample_rate() in snd_usb_endpoint_configure(). Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management") Link: https://lore.kernel.org/r/20210118075816.25068-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Annotate the endpoint index in audioformatTakashi Iwai2021-01-081-1/+1
| | | | | | | | | | | | | | There are devices that have multiple endpoints sharing the same iface/altset not only for sync but also for the actual streams, and the audioformat for such an endpoint needs to be handled with the proper endpoint index; otherwise it confuses the endpoint management. This patch extends the audioformat to annotate the endpoint index, and put the proper ep_idx=1 to Pioneer device quirk entries accordingly. Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management") Link: https://lore.kernel.org/r/20210108075219.21463-5-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Avoid unnecessary interface re-setupTakashi Iwai2021-01-081-11/+71
| | | | | | | | | | | | | | | | | | | | | | | | | The current endpoint handling assumed (more or less) a unique 1:1 relation between the endpoint and the iface/altset. The exception was the sync EP without the implicit feedback which has usually the secondary EP of the same altset. This works fine for most devices, but it turned out that some unusual devices like Pinoeer's ones have both playback and capture endpoints in the same iface/altsetting and use both for the implicit feedback mode. For handling such a case, we need to extend the endpoint management to take the shared interface into account. This patch does that: it adds a new object snd_usb_iface_ref for managing the reference counts of the each USB interface that is used by each endpoint. The interface setup is performed only once for the (sharing) endpoints, and the doubly initialization is avoided. Along with this, the resource release of endpoints and interface refcounts are put into a single function, snd_usb_endpoint_free_all() instead of looping in the caller side. Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management") Link: https://lore.kernel.org/r/20210108075219.21463-4-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Use unsigned char for iface and altsettings fieldsTakashi Iwai2020-11-231-1/+1
| | | | | | | | | | | Just for consistency, use unsigned char for iface and altsetting in allover places. Also rearrange the field positions of snd_usb_endpiont and tidy up with some comments. Tested-by: Keith Milner <kamilner@superlative.org> Tested-by: Dylan Robinson <dylan_robinson@motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-35-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Replace slave/master termsTakashi Iwai2020-11-231-14/+14
| | | | | | | | | | | Follow the inclusive terminology, just replace sync_master/sync_slave with sync_source/sync_sink. It's also a bit clearer from its meaning, too. Tested-by: Keith Milner <kamilner@superlative.org> Tested-by: Dylan Robinson <dylan_robinson@motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-34-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Unify the code for the next packet size calculationTakashi Iwai2020-11-231-16/+24
| | | | | | | | | | | There are two places calculating the next packet size for the playback stream in the exactly same way. Provide the single helper for this purpose and use it from both places gracefully. Tested-by: Keith Milner <kamilner@superlative.org> Tested-by: Dylan Robinson <dylan_robinson@motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-32-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Refactoring endpoint URB deactivationTakashi Iwai2020-11-231-18/+23
| | | | | | | | | | | | Minor code refactoring to consolidate the URB deactivation code in endpoint.c. A slight behavior change is that the error handling in snd_usb_endpoint_start() leaves EP_FLAG_STOPPING now. This should be synced with the later PCM sync_stop callback. Tested-by: Keith Milner <kamilner@superlative.org> Tested-by: Dylan Robinson <dylan_robinson@motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-30-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Use atomic_t for endpoint use_countTakashi Iwai2020-11-231-12/+14
| | | | | | | | | | | | | | | | | | The endpoint objects may be started/stopped concurrently by different substreams in the case of implicit feedback mode, while the current code handles the reference counter without any protection. This patch changes the refcount to atomic_t for avoiding the inconsistency. We need no reference_t here as the refcount goes only up to 2. Also the name "use_count" is renamed to "running" since this is about actually the running status, not the open refcount. Tested-by: Keith Milner <kamilner@superlative.org> Tested-by: Dylan Robinson <dylan_robinson@motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-29-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Constify audioformat pointer referencesTakashi Iwai2020-11-231-1/+1
| | | | | | | | | | The audioformat is referred in many places but most of usages are read-only. Let's add const prefix in the possible places. Tested-by: Keith Milner <kamilner@superlative.org> Tested-by: Dylan Robinson <dylan_robinson@motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-28-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Fix possible stall of implicit fb packet ring-bufferTakashi Iwai2020-11-231-23/+60
| | | | | | | | | | | | | | | | | | | | | The implicit feedback mode uses a ring buffer for storing the received packet sizes from the feedback source, and the code has a slight flaw; when a playback stream stalls by some reason and the URBs aren't processed, the next_packet FIFO might become empty, but the driver can't distinguish whether it's empty or full because it's managed with read_poss and write_pos. This patch addresses those by changing the next_packet array management. Instead of keeping read and write positions, now the head position and the queued amount are kept. It's easier to understand about the emptiness. Also, the URB active flag is now cleared before calling queue_pending_output_urbs() for avoiding (theoretically) possible inconsistency. Tested-by: Keith Milner <kamilner@superlative.org> Tested-by: Dylan Robinson <dylan_robinson@motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-27-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Refactor endpoint managementTakashi Iwai2020-11-231-326/+336
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This is an intensive surgery for the endpoint and stream management for achieving more robust and clean code. The goals of this patch are: - More clear endpoint resource changes - The interface altsetting control in a single place Below are brief description of the whole changes. First off, most of the endpoint operations are moved into endpoint.c, so that the snd_usb_endpoint object is only referred in other places. The endpoint object is acquired and released via the new functions snd_usb_endpoint_open() and snd_usb_endpoint_close() that are called at PCM hw_params and hw_free callbacks, respectively. Those are ref-counted and EPs can manage the multiple opens. The open callback receives the audioformat and hw_params arguments, and those are used for initializing the EP parameters; especially the endpoint, interface and altset numbers are read from there, as well as the PCM parameters like the format, rate and channels. Those are stored in snd_usb_endpoint object. If it's the secondary open, the function checks whether the given parameters are compatible with the already opened EP setup, too. The coupling with a sync EP (including an implicit feedback sync) is done by the sole snd_usb_endpoint_set_sync() call. The configuration of each endpoint is done in a single shot via snd_usb_endpoint_configure() call. This is the place where most of PCM configurations are done. A few flags and special handling in the snd_usb_substream are dropped along with this change. A significant difference wrt the configuration from the previous code is the order of USB host interface setups. Now the interface is always disabled at beginning and (re-)enabled at the last step of snd_usb_endpoint_configure(), in order to be compliant with the standard UAC2/3. For UAC1, the interface is set before the parameter setups since there seem devices that require it (e.g. Yamaha THR10), just like how it was done in the previous driver code. The start/stop are almost same as before, also single-shots. The URB callbacks need to be set via snd_usb_endpoint_set_callback() like the previous code at the trigger phase, too. Finally, the flag for the re-setup is set at the device suspend through the full EP list, instead of PCM trigger. This catches the overlooked cases where the PCM hasn't been running yet but the device needs the full setup after resume. Tested-by: Keith Milner <kamilner@superlative.org> Tested-by: Dylan Robinson <dylan_robinson@motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-26-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Set callbacks via snd_usb_endpoint_set_callback()Takashi Iwai2020-11-231-19/+41
| | | | | | | | | | | | | | | | | | | | | | | | | The prepare_data_urb and retire_data_urb fields of the endpoint object are set dynamically at PCM trigger start/stop. Those are evaluated in the endpoint handler, but there can be a race, especially if two different PCM substreams are handling the same endpoint for the implicit feedback case. Also, the data_subs field of the endpoint is set and accessed dynamically, too, which has the same risk. As a slight improvement for the concurrency, this patch introduces the function to set the callbacks and the data in a shot with the memory barrier. In the reader side, it's also fetched with the memory barrier. There is still a room of race if prepare and retire callbacks are set during executing the URB completion. But such an inconsistency may happen only for the implicit fb source, i.e. it's only about the capture stream. And luckily, the capture stream never sets the prepare callback, hence the problem doesn't happen practically. Tested-by: Keith Milner <kamilner@superlative.org> Tested-by: Dylan Robinson <dylan_robinson@motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-23-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Stop both endpoints properly at errorTakashi Iwai2020-11-231-0/+6
| | | | | | | | | | | | | | start_endpoints() may leave the data endpoint running if an error happens at starting the sync endpoint. We should stop both streams properly, instead. While we're at it, move the debug prints into the endpoint.c that is a more suitable place. Tested-by: Keith Milner <kamilner@superlative.org> Tested-by: Dylan Robinson <dylan_robinson@motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-22-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Create endpoint objects at parsing phaseTakashi Iwai2020-11-231-53/+34
| | | | | | | | | | | | | | | | | | | Currently snd_usb_endpoint objects are created at first when the substream is opened and tries to assign the endpoints corresponding to the matching audioformat. But since basically the all endpoints have been already parsed and the information have been obtained, we may create the endpoint objects statically at the init phase. It's easier to manage for the implicit fb case, for example. This patch changes the endpoint object management and lets the parser to create the all endpoint objects. This change shouldn't bring any functional changes. Tested-by: Keith Milner <kamilner@superlative.org> Tested-by: Dylan Robinson <dylan_robinson@motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-15-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Add hw constraint for implicit fb syncTakashi Iwai2020-11-231-19/+53
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | In the current code, there is no check at the stream open time whether the endpoint is being already used by others. In the normal operations, this shouldn't happen, but in the case of the implicit feedback mode, it's a common problem with the full duplex operation, because the capture stream is always opened by the playback stream as an implicit sync source. Although we recently introduced the check of such a conflict of parameters at the PCM hw_params time, it doesn't give any hint at the hw_params itself and just gives the error. This isn't quite comfortable, and it caused problems on many applications. This patch attempts to make the parameter handling easier by introducing the strict hw constraint matching with the counterpart stream that is being used. That said, when an implicit feedback playback stream is running before a capture stream is opened, the capture stream carries the PCM hw-constraint to allow only the same sample rate, format, periods and period frames as the running playback stream. If not opened or there is no conflict of endpoints, the behavior remains as same as before. Note that this kind of "weak link" should work for most cases, but this is no concrete solution; e.g. if an application changes the hw params multiple times while another stream is opened, this would lead to inconsistencies. Tested-by: Keith Milner <kamilner@superlative.org> Tested-by: Dylan Robinson <dylan_robinson@motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-11-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Improve some debug printsTakashi Iwai2020-11-231-4/+4
| | | | | | | | | | | | | | There are a few rooms for improvements wrt the debug prints: - The EP debug print is shown only at starting, not at stopping - The EP debug print contains useless object addresses - Some helpers show the urb and the EP object addresses, too This patch addresses those shortcomings. Tested-by: Keith Milner <kamilner@superlative.org> Tested-by: Dylan Robinson <dylan_robinson@motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-8-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Add snd_usb_get_endpoint() helperTakashi Iwai2020-11-231-9/+27
| | | | | | | | | | | | | Factor out the code to obtain snd_usb_endpoint object matching with the given endpoint. It'll be used in the later patch to add the implicit feedback hw-constraint. No functional change by this patch itself. Tested-by: Keith Milner <kamilner@superlative.org> Tested-by: Dylan Robinson <dylan_robinson@motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-6-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
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