This adds a button on the menubar next to the volume slider to
indicate mute state and allow toggling the mute. Pressing the M key
will still toggle the mute, as before. When muted, the volume scroll
bar now gets disabled.
When the visualization is set to "Album Cover", the player will now try
to load the embedded image. On failure, it defaults to a "Cover" image
file in the directory.
In Player::play_file_path, file_name_changed now needs to be executed
after that the loader have been set, to get the correct image.
This patch will switch cursor to DragCopy when a user enters a widget
while dragging file(s), giving them a visual clue that it *might* be
dropped into this widget.
This is a rather naive approach, as the cursor icon will change for any
kind of file, as currently programs don't know the drag contents before
dropping it. But after all I think it's better than nothing. :^)
Previously, we were sending Buffers to the server whenever we had new
audio data for it. This meant that for every audio enqueue action, we
needed to create a new shared memory anonymous buffer, send that
buffer's file descriptor over IPC (+recfd on the other side) and then
map the buffer into the audio server's memory to be able to play it.
This was fine for sending large chunks of audio data, like when playing
existing audio files. However, in the future we want to move to
real-time audio in some applications like Piano. This means that the
size of buffers that are sent need to be very small, as just the size of
a buffer itself is part of the audio latency. If we were to try
real-time audio with the existing system, we would run into problems
really quickly. Dealing with a continuous stream of new anonymous files
like the current audio system is rather expensive, as we need Kernel
help in multiple places. Additionally, every enqueue incurs an IPC call,
which are not optimized for >1000 calls/second (which would be needed
for real-time audio with buffer sizes of ~40 samples). So a fundamental
change in how we handle audio sending in userspace is necessary.
This commit moves the audio sending system onto a shared single producer
circular queue (SSPCQ) (introduced with one of the previous commits).
This queue is intended to live in shared memory and be accessed by
multiple processes at the same time. It was specifically written to
support the audio sending case, so e.g. it only supports a single
producer (the audio client). Now, audio sending follows these general
steps:
- The audio client connects to the audio server.
- The audio client creates a SSPCQ in shared memory.
- The audio client sends the SSPCQ's file descriptor to the audio server
with the set_buffer() IPC call.
- The audio server receives the SSPCQ and maps it.
- The audio client signals start of playback with start_playback().
- At the same time:
- The audio client writes its audio data into the shared-memory queue.
- The audio server reads audio data from the shared-memory queue(s).
Both sides have additional before-queue/after-queue buffers, depending
on the exact application.
- Pausing playback is just an IPC call, nothing happens to the buffer
except that the server stops reading from it until playback is
resumed.
- Muting has nothing to do with whether audio data is read or not.
- When the connection closes, the queues are unmapped on both sides.
This should already improve audio playback performance in a bunch of
places.
Implementation & commit notes:
- Audio loaders don't create LegacyBuffers anymore. LegacyBuffer is kept
for WavLoader, see previous commit message.
- Most intra-process audio data passing is done with FixedArray<Sample>
or Vector<Sample>.
- Improvements to most audio-enqueuing applications. (If necessary I can
try to extract some of the aplay improvements.)
- New APIs on LibAudio/ClientConnection which allows non-realtime
applications to enqueue audio in big chunks like before.
- Removal of status APIs from the audio server connection for
information that can be directly obtained from the shared queue.
- Split the pause playback API into two APIs with more intuitive names.
I know this is a large commit, and you can kinda tell from the commit
message. It's basically impossible to break this up without hacks, so
please forgive me. These are some of the best changes to the audio
subsystem and I hope that that makes up for this :yaktangle: commit.
:yakring:
With the following change in how we send audio, the old Buffer type is
not really needed anymore. However, moving WavLoader to the new system
is a bit more involved and out of the scope of this PR. Therefore, we
need to keep Buffer around, but to make it clear that it's the old
buffer type which will be removed soon, we rename it to LegacyBuffer.
Most of the users will be gone after the next commit anyways.
This adds a new start_new_file() function to VisualizationWidget which
is called when the player starts a new file, passing the filename of the
file. This allows VisualizationWidget subclasses to do any setup needed
when a new file is started.
Derivatives of Core::Object should be constructed through
ClassName::construct(), to avoid handling ref-counted objects with
refcount zero. Fixing the visibility means that misuses like this are
more difficult.
The shuffling algorithm uses a naïve bloom filter to provide random
uniformity, avoiding items that were recently played. With 32 bits,
double hashing, and an error rate of ~10%, this bloom filter should
be able to hold around ~16 keys, which should be sufficient to give the
illusion of fairness to the shuffling algorithm.
This avoids having to shuffle the playlist itself (user might have
spent quite a bit of time to sort them, so it's not a good idea to mess
with it), or having to create a proxy model that shuffles (that could
potentially use quite a bit of memory).
This is a first pass at refactoring SoundPlayer so that the View widget
is decoupled from the player itself.
In doing so, this fixed a couple of issues, including possibly
inconsistent states (e.g. player could be paused and stopped at the
same time).
With the change, Player actually controls the show, and calls methods
overriden by its subclasses to perform actions, such as update the Seek
bar; the hard work of massaging the raw data is done by the Player
class, so subclasses don't need to reimplement any of these things.
This also removes some copies of playlist management code that happened
to be copied+pasted inside callbacks of buttons -- it now lives inside
a neatly packaged Playlist class, and the Player only asks for the next
song to play.
In addition, the menu bar has been slightly rearranged.
This commit addresses two issues:
1. If you play a 96 KHz Wave file, the slider position is incorrect,
because it is assumed all files are 44.1 KHz.
2. For high-bitrate files, there are audio dropouts due to not
buffering enough audio data.
Issue 1 is addressed by scaling the number of played samples by the
ratio between the source and destination sample rates.
Issue 2 is addressed by buffering a certain number of milliseconds
worth of audio data (instead of a fixed number of bytes).
This makes the the buffer size independent of the source sample rate.
Some of the code is redesigned to be simpler. The code that did the
book-keeping of which buffers need to be loaded and which have been
already played has been removed. Instead, we enqueue a new buffer based
on a low watermark of samples remaining in the audio server queue.
Other small fixes include:
1. Disable the stop button when playback is finished.
2. Remove hard-coded instances of 44100.
3. Update the GUI every 50 ms (was 100), which improves visualizations.
SPDX License Identifiers are a more compact / standardized
way of representing file license information.
See: https://spdx.dev/resources/use/#identifiers
This was done with the `ambr` search and replace tool.
ambr --no-parent-ignore --key-from-file --rep-from-file key.txt rep.txt *
