This feels a lot more consistent and Unixy:
create_shared_buffer() => shbuf_create()
share_buffer_with() => shbuf_allow_pid()
share_buffer_globally() => shbuf_allow_all()
get_shared_buffer() => shbuf_get()
release_shared_buffer() => shbuf_release()
seal_shared_buffer() => shbuf_seal()
get_shared_buffer_size() => shbuf_get_size()
Also, "shared_buffer_id" is shortened to "shbuf_id" all around.
This class mirrors WavLoader and uses the same variable names so they
can be read side-by-side.
In a simple use-case, you construct a WavWriter object with your file
path and audio parameters, call write_samples() with your bytes and then
either finalize() to finish writing the file or allow the destructor to
do it automatically.
finalize() is needed because the WAV header cannot be written until the
size of the data section is known. You only need to call this manually
if you want to write a WAV file and then immediately load it.
You can also use one WavWriter instance multiple times by repeatedly
calling finalize() and set_file().
I've been wanting to do this for a long time. It's time we start being
consistent about how this stuff works.
The new convention is:
- "LibFoo" is a userspace library that provides the "Foo" namespace.
That's it :^) This was pretty tedious to convert and I didn't even
start on LibGUI yet. But it's coming up next.
As suggested by Joshua, this commit adds the 2-clause BSD license as a
comment block to the top of every source file.
For the first pass, I've just added myself for simplicity. I encourage
everyone to add themselves as copyright holders of any file they've
added or modified in some significant way. If I've added myself in
error somewhere, feel free to replace it with the appropriate copyright
holder instead.
Going forward, all new source files should include a license header.
Lock each directory before entering it so when using -j, the same
dependency isn't built more than once at a time.
This doesn't get full -j parallelism though, since one make child
will be sitting idle waiting for flock to receive its lock and
continue making (which should then do nothing since it will have
been built already). Unfortunately there's not much that can be
done to fix that since it can't proceed until its dependency is
built by another make process.
Instead of directly manipulating LDFLAGS, set LIB_DEPS in each
subdirectory Makefile listing the libraries needed for
building/linking such as "LIB_DEPS = Core GUI Draw IPC Core".
This adds each library as an -L and -l argument in LDFLAGS, but
also adds the library.a file as a link dependency on the current
$(PROGRAM). This causes the given library to be (re)built before
linking the current $(PROGRAM), but will also re-link any binaries
depending on that library when it is modified, when running make
from the root directory.
Also turn generator tools like IPCCompiler into dependencies on the
files they generate, so they are built on-demand when a particular
directory needs them.
This all allows the root Makefile to just list directories and not
care about the order, as all of the dependency tracking will figure
it out.
Allow everything to be built from the top level directory with just
'make', cleaned with 'make clean', and installed with 'make
install'. Also support these in any particular subdirectory.
Specifying 'make VERBOSE=1' will print each ld/g++/etc. command as
it runs.
Kernel and early host tools (IPCCompiler, etc.) are built as
object.host.o so that they don't conflict with other things built
with the cross-compiler.
Instead of passing the PIDs back and forth in a handshake "Greet"
message, just use getsockopt(SO_PEERCRED) on both sides to get the same
information from the kernel.
This is a nice little simplification of the IPC protocol, although it
does not get rid of the handshake since we still have to pass the
"client ID" from the server to each client so they know how to refer
to themselves. This might not be necessary and we might be able to get
rid of this later on.
This matches what we're already calling the server-side subclasses
better, though we'll probably want to find some better names for the
client-side classes eventually.
Client-side connection objects must now provide both client and server
endpoint types. When a message is received from the server side, we try
to decode it using both endpoint types and then send it to the right
place for handling.
This now makes it possible for AudioServer to send unsolicited messages
to its clients. This opens up a ton of possibilities :^)
This patch adds muting to ASMixer, which works by substituting what we
would normally send to the sound card with zero-filled memory instead.
We do it this way to ensure that the queued sample buffers keep getting
played (silently.)
This is obviously not the perfect way of doing this, and in the future
we should improve on this, and also find a way to utilize any hardware
mixing functions in the sound card.
Renamed "Position" to "Elapsed". "channel/channels" automatically
changes now when more than one channel exist. The current file name
is now displayed in the window title.
m_loaded_samples was incremented with the value of the processed
buffer. This causes m_loaded_samples to be bigger at some point
than m_total_samples when downsampling, as the buffer would contain
more samples than actually loaded.
LibAudio now supports pausing playback, clearing the buffer queue,
retrieving the played samples since the last clear and retrieving
the currently playing shared buffer id
When playing an ABuffer, the count of samples were determined by the
size of the SharedBuffer. This caused small pauses of up to 512
samples during the playback, when the size of the shared buffer was
rounded up to a multiple of 4096. This problem was amplified by the
fact that the AResampleHelper was created every time a new chunk of
audio was to be processed, causing inconsistencies in the playback of
wav files.
This was a workaround to be able to build on case-insensitive file
systems where it might get confused about <string.h> vs <String.h>.
Let's just not support building that way, so String.h can have an
objectively nicer name. :^)
This is a variant of the enqueue() API that returns immediately and
may fail. It's useful when you don't want to block until the audio
server can receive your sample buffer.
Fork the IPC Connection classes into Server:: and Client::ConnectionNG.
The new IPC messages are serialized very snugly instead of using the
same generic data structure for all messages.
Remove ASAPI.h since we now generate all of it from AudioServer.ipc :^)
The goal here is to generate most of this code from IPC protocol
descriptions, but for now I've spelled them all out to get started.
Each message gets a wrapper class in the ASAPI_Client or ASAPI_Server
namespace. They are convertible to and from the old message structs.
The real hotness happens when you want to make a synchronous request
to the other side:
auto response = send_sync<ASAPI_Client::GetMainMixVolume>();
Each request class knows his corresponding response class, so in the
above example, "response" will be an ASAPI_Server::DidGetMainMixVolume
object, and we can get the volume like so:
int volume = response.volume();
For posting messages that don't expect a response, you can still use
post_message() since the message classes are convertible:
post_message(ASAPI_Server::DidGetMainMixVolume(volume));
It's not perfect yet, but I already really like it. :^)
Give the mixer a main volume value (percent) that we scale all the
outgoing samples by (before clipping.)
Also add a simple "avol" program for querying and setting the volume:
- "avol" prints the current volume.
- "avol 200" sets the main mix volume to 200%
Show some information about the file we're playing, and display how many
samples we've played out of how many total.
This might be a bit buggy as I haven't tested it with many different files,
but it's a start. :^)
This is a total hack, because I haven't really looked into why these are
happening. Somehow we're producing one extra sample and it's glitching
up the sound stream ever so slightly.
Each client connection now sets up an ASBufferQueue, which is basically a
queue of ABuffers. This allows us to immediately start streaming the next
pending buffer whenever our current buffer runs out of samples.
This makes the majority of the skippiness go away for me. :^)
Also get rid of the old PlayBuffer API, since we don't need it anymore.
We were limiting ourselves to only play WAV files smaller than 42 MB
for no particular reason. This patch increases the limit to 1 GB.
Perhaps there should not be any limit at all, but 1GB seems like a
reasonable sanity check at the moment. :^)
