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LibAudio: New error propagation API in Loader and Buffer

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Previously, a libc-like out-of-line error information was used in the
loader and its plugins. Now, all functions that may fail to do their job
return some sort of Result. The universally-used error type ist the new
LoaderError, which can contain information about the general error
category (such as file format, I/O, unimplemented features), an error
description, and location information, such as file index or sample
index.

Additionally, the loader plugins try to do as little work as possible in
their constructors. Right after being constructed, a user should call
initialize() and check the errors returned from there. (This is done
transparently by Loader itself.) If a constructor caused an error, the
call to initialize should check and return it immediately.

This opportunity was used to rework a lot of the internal error
propagation in both loader classes, especially FlacLoader. Therefore, a
couple of other refactorings may have sneaked in as well.

The adoption of LibAudio users is minimal. Piano's adoption is not
important, as the code will receive major refactoring in the near future
anyways. SoundPlayer's adoption is also less important, as changes to
refactor it are in the works as well. aplay's adoption is the best and
may serve as an example for other users. It also includes new buffering
behavior.

Buffer also gets some attention, making it OOM-safe and thereby also
propagating its errors to the user.
This commit is contained in:
kleines Filmröllchen 2021-11-27 17:00:19 +01:00 • committed by Brian Gianforcaro
parent ec8bd8116d
commit 96d02a3e75
19 changed files with 406 additions and 397 deletions
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64
Userland/Utilities/aplay.cpp
View file

@ -4,6 +4,7 @@
* SPDX-License-Identifier: BSD-2-Clause
*/
#include <AK/Types.h>
#include <LibAudio/ClientConnection.h>
#include <LibAudio/Loader.h>
#include <LibCore/ArgsParser.h>
@ -11,6 +12,10 @@
#include <LibMain/Main.h>
#include <stdio.h>
// The Kernel has issues with very large anonymous buffers.
// FIXME: This appears to be fine for now, but it's really a hack.
constexpr size_t LOAD_CHUNK_SIZE = 128 * KiB;
ErrorOr<int> serenity_main(Main::Arguments arguments)
{
const char* path = nullptr;
@ -24,11 +29,12 @@ ErrorOr<int> serenity_main(Main::Arguments arguments)
Core::EventLoop loop;
auto audio_client = Audio::ClientConnection::construct();
NonnullRefPtr<Audio::Loader> loader = Audio::Loader::create(path);
if (loader->has_error()) {
warnln("Failed to load audio file: {}", loader->error_string());
auto maybe_loader = Audio::Loader::create(path);
if (maybe_loader.is_error()) {
warnln("Failed to load audio file: {}", maybe_loader.error().description);
return 1;
}
auto loader = maybe_loader.release_value();
outln("\033[34;1m Playing\033[0m: {}", path);
outln("\033[34;1m Format\033[0m: {} Hz, {}-bit, {}",
@ -39,25 +45,43 @@ ErrorOr<int> serenity_main(Main::Arguments arguments)
auto resampler = Audio::ResampleHelper<double>(loader->sample_rate(), audio_client->get_sample_rate());
// If we're downsampling, we need to appropriately load more samples at once.
size_t const load_size = static_cast<size_t>(LOAD_CHUNK_SIZE * static_cast<double>(loader->sample_rate()) / static_cast<double>(audio_client->get_sample_rate()));
// We assume that the loader can load samples at at least 2x speed (testing confirms 9x-12x for FLAC, 14x for WAV).
// Therefore, when the server-side buffer can only play as long as the time it takes us to load a chunk,
// we give it new data.
int const min_buffer_size = load_size / 2;
for (;;) {
auto samples = loader->get_more_samples();
if (samples) {
out("\033[u");
out("{}/{}", loader->loaded_samples(), loader->total_samples());
fflush(stdout);
resampler.reset();
samples = Audio::resample_buffer(resampler, *samples);
audio_client->enqueue(*samples);
} else if (loader->has_error()) {
outln();
outln("Error: {}", loader->error_string());
break;
} else if (should_loop) {
loader->reset();
} else if (audio_client->get_remaining_samples()) {
sleep(1);
auto samples = loader->get_more_samples(load_size);
if (!samples.is_error()) {
if (samples.value()->sample_count() > 0) {
// We can read and enqueue more samples
out("\033[u");
out("{}/{}", loader->loaded_samples(), loader->total_samples());
fflush(stdout);
resampler.reset();
auto resampled_samples = TRY(Audio::resample_buffer(resampler, *samples.value()));
audio_client->async_enqueue(*resampled_samples);
} else if (should_loop) {
// We're done: now loop
auto result = loader->reset();
if (result.is_error()) {
outln();
outln("Error while resetting: {} (at {:x})", result.error().description, result.error().index);
}
} else if (samples.value()->sample_count() == 0 && audio_client->get_remaining_samples() == 0) {
// We're done and the server is done
break;
}
while (audio_client->get_remaining_samples() > min_buffer_size) {
// The server has enough data for now
sleep(1);
}
} else {
break;
outln();
outln("Error: {} (at {:x})", samples.error().description, samples.error().index);
return 1;
}
}
outln();