2021-09-30 04:55:42 +00:00
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/*
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* Copyright (c) 2021, Cesar Torres <shortanemoia@protonmail.com>
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* Copyright (c) 2021, Leandro A. F. Pereira <leandro@tia.mat.br>
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*
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* SPDX-License-Identifier: BSD-2-Clause
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*/
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#include "Player.h"
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2022-10-03 23:07:18 +00:00
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#include <LibAudio/FlacLoader.h>
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2023-03-21 15:35:30 +00:00
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#include <LibFileSystem/FileSystem.h>
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2021-09-30 04:55:42 +00:00
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2022-07-17 09:31:01 +00:00
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Player::Player(Audio::ConnectionToServer& audio_client_connection)
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2021-09-30 04:55:42 +00:00
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: m_audio_client_connection(audio_client_connection)
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, m_playback_manager(audio_client_connection)
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{
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m_playback_manager.on_update = [&]() {
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LibAudio+Userland: Use new audio queue in client-server communication
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:
2022-02-20 12:01:22 +00:00
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auto samples_played = m_playback_manager.loader()->loaded_samples();
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2021-09-30 04:55:42 +00:00
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auto sample_rate = m_playback_manager.loader()->sample_rate();
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float source_to_dest_ratio = static_cast<float>(sample_rate) / m_playback_manager.device_sample_rate();
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samples_played *= source_to_dest_ratio;
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auto played_seconds = samples_played / sample_rate;
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time_elapsed(played_seconds);
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sound_buffer_played(m_playback_manager.current_buffer(), m_playback_manager.device_sample_rate(), samples_played);
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};
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m_playback_manager.on_finished_playing = [&]() {
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set_play_state(PlayState::Stopped);
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switch (loop_mode()) {
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case LoopMode::File:
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play_file_path(loaded_filename());
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return;
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case LoopMode::Playlist:
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play_file_path(m_playlist.next());
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return;
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case LoopMode::None:
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return;
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}
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};
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}
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2022-12-04 18:02:33 +00:00
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void Player::play_file_path(DeprecatedString const& path)
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2021-09-30 04:55:42 +00:00
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{
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if (path.is_null())
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return;
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2023-03-21 15:35:30 +00:00
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if (!FileSystem::exists(path)) {
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2022-07-11 17:32:29 +00:00
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audio_load_error(path, "File does not exist"sv);
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2021-09-30 04:55:42 +00:00
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return;
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}
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2021-12-20 02:59:28 +00:00
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if (is_playlist(path)) {
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2021-09-30 04:55:42 +00:00
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playlist_loaded(path, m_playlist.load(path));
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return;
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}
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LibAudio: New error propagation API in Loader and Buffer
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.
2021-11-27 16:00:19 +00:00
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auto maybe_loader = Audio::Loader::create(path);
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if (maybe_loader.is_error()) {
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audio_load_error(path, maybe_loader.error().description);
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2021-09-30 04:55:42 +00:00
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return;
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}
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LibAudio: New error propagation API in Loader and Buffer
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.
2021-11-27 16:00:19 +00:00
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auto loader = maybe_loader.value();
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2021-09-30 04:55:42 +00:00
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m_loaded_filename = path;
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2023-01-17 11:20:41 +00:00
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// TODO: The combination of sample count, sample rate, and sample data should be properly abstracted for the source and the playback device.
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total_samples_changed(loader->total_samples() * (static_cast<float>(loader->sample_rate()) / m_playback_manager.device_sample_rate()));
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2021-09-30 04:55:42 +00:00
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m_playback_manager.set_loader(move(loader));
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2022-10-03 23:07:18 +00:00
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file_name_changed(path);
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2021-09-30 04:55:42 +00:00
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play();
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}
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2022-12-04 18:02:33 +00:00
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bool Player::is_playlist(DeprecatedString const& path)
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2021-12-20 02:59:28 +00:00
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{
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2022-07-11 17:32:29 +00:00
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return (path.ends_with(".m3u"sv, AK::CaseSensitivity::CaseInsensitive)
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|| path.ends_with(".m3u8"sv, AK::CaseSensitivity::CaseInsensitive));
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2021-12-20 02:59:28 +00:00
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}
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2021-09-30 04:55:42 +00:00
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void Player::set_play_state(PlayState state)
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{
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if (m_play_state != state) {
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m_play_state = state;
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play_state_changed(state);
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}
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}
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void Player::set_loop_mode(LoopMode mode)
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{
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if (m_loop_mode != mode) {
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m_loop_mode = mode;
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m_playlist.set_looping(mode == LoopMode::Playlist);
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loop_mode_changed(mode);
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}
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}
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void Player::set_volume(double volume)
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{
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2021-10-31 02:38:12 +00:00
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m_volume = clamp(volume, 0, 1.5);
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2021-09-30 04:55:42 +00:00
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m_audio_client_connection.set_self_volume(m_volume);
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volume_changed(m_volume);
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}
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2021-11-02 01:31:05 +00:00
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void Player::set_mute(bool muted)
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{
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if (m_muted != muted) {
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m_muted = muted;
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m_audio_client_connection.set_self_muted(muted);
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mute_changed(muted);
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}
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}
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2021-09-30 14:41:00 +00:00
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void Player::set_shuffle_mode(ShuffleMode mode)
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{
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if (m_shuffle_mode != mode) {
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m_shuffle_mode = mode;
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m_playlist.set_shuffling(mode == ShuffleMode::Shuffling);
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shuffle_mode_changed(mode);
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}
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}
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2021-09-30 04:55:42 +00:00
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void Player::play()
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{
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m_playback_manager.play();
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set_play_state(PlayState::Playing);
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}
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void Player::pause()
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{
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m_playback_manager.pause();
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set_play_state(PlayState::Paused);
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}
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void Player::toggle_pause()
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{
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bool paused = m_playback_manager.toggle_pause();
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set_play_state(paused ? PlayState::Paused : PlayState::Playing);
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}
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void Player::stop()
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{
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m_playback_manager.stop();
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set_play_state(PlayState::Stopped);
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}
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2021-11-02 01:31:05 +00:00
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void Player::mute()
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{
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set_mute(true);
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}
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void Player::toggle_mute()
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{
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set_mute(!m_muted);
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}
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2021-09-30 04:55:42 +00:00
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void Player::seek(int sample)
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{
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2023-01-17 20:27:13 +00:00
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sample *= (m_playback_manager.device_sample_rate() / static_cast<float>(m_playback_manager.loader()->sample_rate()));
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2021-09-30 04:55:42 +00:00
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m_playback_manager.seek(sample);
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}
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2022-10-03 23:07:18 +00:00
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Vector<Audio::PictureData> const& Player::pictures() const
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{
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return m_playback_manager.loader()->pictures();
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}
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