WASAPI_WaitDevice is used for audio playback and capture, but needs to
behave slighty different.
For playback `GetCurrentPadding` returns the padding which is already
queued, so WaitDevice should return when buffer length falls below the
buffer threshold (`maxpadding`).
For capture `GetCurrentPadding` returns the available data which can be
read, so WaitDevice can return as soon as any data is available.
In the old implementation WaitDevice could suddenly hang. This is
because on many capture devices the buffer (`padding`) wasn't filled
fast enough to surpass `maxpadding`. But if at one point (due to unlucky
timing) more than maxpadding frames were available, WaitDevice would not
return anymore.
Issue #3234 is probably related to this.
On modern CPUs, there's no penalty for using the unaligned instruction on
aligned memory, but now it can vectorize unaligned data too, which even if
it's not optimal, is still going to be faster than the scalar fallback.
Fixes#4532.
While we should normally expect _something_ from the stream based on the
AudioStreamAvailable check, it's possible for a device change to flush the
stream at an inconvenient time, causing this function to return 0.
Thing is, this is harmless. Either data will be NULL and the result won't matter
anyway, or the data buffer will be zeroed out and the output will just be
silence for the brief moment that the device change is occurring. Both scenarios
work themselves out, and testing on Windows shows that this behavior is safe.
Some of the SDL_AudioDevice struct members aren't initialized until after returning from the OpenDevice function. Since Pipewire uses it's own processing threads, the callbacks can be entered before all members of SDL_AudioDevice are initialized, such as work_buffer, callbackspec and the processing stream, which creates a race condition. Don't use these members when in the paused state to avoid potentially using uninitialized values and memory.
This prevents the dsp target from stealing the audio subsystem but not
being able to produce sound, so other audio targets further down the list
can make an attempt instead.
Thanks to Frank Praznik who did a lot of the research on this problem!
A user reported that the mpv video player hangs after attempting to
set an unsupported number of channels with the SDL audio output,
because it thinks it's successfully opened the device. This makes
the failure graceful.
Removes the node nickname from sink/source nodes as it doesn't provide any useful information and names now match those used in Pulseaudio, so any stored configuration data will be compatible between the two audio backends.
Constify the min/max period variables, use a #define for the base clock rate used in the calculations and note that changing the upper limit can have dire side effects as it's a hard limit in Pipewire.
Replace "magic numbers" with #defines, explain the requirements when using the userdata pointer in the node_object struct and a few other minor code and comment cleanups.
Use the 'R' (rear) prefixed designations for the rear audio channels instead of 'S' (surround). Surround designated channels are only used in the 8 channel configuration.
Further refactor the device enumeration code to retrieve the default sink/source node IDs from the metadata node. Use the retrieved IDs to sort the device list so that the default devices are at the beginning and thus are the first reported to SDL.
The latency of source nodes can change depending on the overall latency of the processing graph. Incoming audio must therefore always be buffered to ensure uninterrupted delivery.
The SDL_AudioStream path was removed in the input callback as the only thing it was used for was buffering audio outside of Pipewire's min/max period sizes, and that case is now handled by the omnipresent buffer.
