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https://github.com/LadybirdBrowser/ladybird.git
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b4fbd30b70
This change was a long time in the making ever since we obtained sample rate awareness in the system. Now, each client has its own sample rate, accessible via new IPC APIs, and the device sample rate is only accessible via the management interface. AudioServer takes care of resampling client streams into the device sample rate. Therefore, the main improvement introduced with this commit is full responsiveness to sample rate changes; all open audio programs will continue to play at correct speed with the audio resampled to the new device rate. The immediate benefits are manifold: - Gets rid of the legacy hardware sample rate IPC message in the non-managing client - Removes duplicate resampling and sample index rescaling code everywhere - Avoids potential sample index scaling bugs in SoundPlayer (which have happened many times before) and fixes a sample index scaling bug in aplay - Removes several FIXMEs - Reduces amount of sample copying in all applications (especially Piano, where this is critical), improving performance - Reduces number of resampling users, making future API changes (which will need to happen for correct resampling to be implemented) easier I also threw in a simple race condition fix for Piano's audio player loop.
121 lines
4.7 KiB
C++
121 lines
4.7 KiB
C++
/*
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* Copyright (c) 2018-2020, Andreas Kling <kling@serenityos.org>
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* Copyright (c) 2021-2022, kleines Filmröllchen <filmroellchen@serenityos.org>
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*
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* SPDX-License-Identifier: BSD-2-Clause
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*/
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#include <AK/Types.h>
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#include <LibAudio/ConnectionToServer.h>
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#include <LibAudio/Loader.h>
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#include <LibCore/ArgsParser.h>
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#include <LibCore/EventLoop.h>
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#include <LibCore/System.h>
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#include <LibFileSystem/FileSystem.h>
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#include <LibMain/Main.h>
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#include <math.h>
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#include <stdio.h>
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constexpr size_t LOAD_CHUNK_SIZE = 128 * KiB;
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ErrorOr<int> serenity_main(Main::Arguments arguments)
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{
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TRY(Core::System::pledge("stdio rpath sendfd unix thread proc"));
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StringView path {};
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bool should_loop = false;
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bool show_sample_progress = false;
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Core::ArgsParser args_parser;
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args_parser.add_positional_argument(path, "Path to audio file", "path");
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args_parser.add_option(should_loop, "Loop playback", "loop", 'l');
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args_parser.add_option(show_sample_progress, "Show playback progress in samples", "sample-progress", 's');
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args_parser.parse(arguments);
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// Note: We must determine the absolute path *before* beginning to raise the veil.
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auto absolute_path = TRY(FileSystem::absolute_path(path));
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TRY(Core::System::unveil("/tmp/session/%sid/portal/audio", "rw"));
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TRY(Core::System::unveil(absolute_path, "r"sv));
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TRY(Core::System::unveil(nullptr, nullptr));
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Core::EventLoop loop;
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auto audio_client = TRY(Audio::ConnectionToServer::try_create());
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auto maybe_loader = Audio::Loader::create(path);
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if (maybe_loader.is_error()) {
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warnln("Failed to load audio file: {}", maybe_loader.error().description);
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return 1;
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}
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auto loader = maybe_loader.release_value();
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TRY(Core::System::pledge("stdio sendfd thread proc"));
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outln("\033[34;1m Playing\033[0m: {}", path);
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outln("\033[34;1m Format\033[0m: {} {} Hz, {}-bit, {}",
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loader->format_name(),
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loader->sample_rate(),
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loader->bits_per_sample(),
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loader->num_channels() == 1 ? "Mono" : "Stereo");
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out("\033[34;1mProgress\033[0m: \033[s");
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audio_client->set_self_sample_rate(loader->sample_rate());
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auto print_playback_update = [&]() {
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out("\033[u");
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if (show_sample_progress) {
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out("{}/{}", audio_client->total_played_samples(), loader->total_samples());
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} else {
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auto playing_seconds = static_cast<int>(floor(static_cast<double>(audio_client->total_played_samples()) / static_cast<double>(loader->sample_rate())));
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auto playing_minutes = playing_seconds / 60;
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auto playing_seconds_of_minute = playing_seconds % 60;
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auto total_seconds = static_cast<int>(floor(static_cast<double>(loader->total_samples()) / static_cast<double>(loader->sample_rate())));
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auto total_minutes = total_seconds / 60;
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auto total_seconds_of_minute = total_seconds % 60;
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auto remaining_seconds = total_seconds - playing_seconds;
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auto remaining_minutes = remaining_seconds / 60;
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auto remaining_seconds_of_minute = remaining_seconds % 60;
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out("\033[1m{:02d}:{:02d}\033[0m [{}{:02d}:{:02d}] -- {:02d}:{:02d}",
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playing_minutes, playing_seconds_of_minute,
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remaining_seconds == 0 ? " " : "-",
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remaining_minutes, remaining_seconds_of_minute,
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total_minutes, total_seconds_of_minute);
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}
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fflush(stdout);
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};
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for (;;) {
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auto samples = loader->get_more_samples(LOAD_CHUNK_SIZE);
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if (!samples.is_error()) {
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if (samples.value().size() > 0) {
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print_playback_update();
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// We can read and enqueue more samples
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TRY(audio_client->async_enqueue(samples.release_value()));
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} else if (should_loop) {
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// We're done: now loop
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auto result = loader->reset();
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if (result.is_error()) {
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outln();
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outln("Error while resetting: {} (at {:x})", result.error().description, result.error().index);
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}
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} else if (samples.value().size() == 0 && audio_client->remaining_samples() == 0) {
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// We're done and the server is done
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break;
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}
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while (audio_client->remaining_samples() > LOAD_CHUNK_SIZE) {
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// The server has enough data for now
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print_playback_update();
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usleep(1'000'000 / 10);
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}
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} else {
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outln();
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outln("Error: {} (at {:x})", samples.error().description, samples.error().index);
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return 1;
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}
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}
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outln();
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return 0;
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}
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