initial commit

Signed-off-by: Sagi Dayan <sagidayan@gmail.com>

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LICENSE

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+                    GNU GENERAL PUBLIC LICENSE
+                       Version 3, 29 June 2007
+
+ Copyright (C) 2007 Free Software Foundation, Inc. <https://fsf.org/>
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+  16. Limitation of Liability.
+
+  IN NO EVENT UNLESS REQUIRED BY APPLICABLE LAW OR AGREED TO IN WRITING
+WILL ANY COPYRIGHT HOLDER, OR ANY OTHER PARTY WHO MODIFIES AND/OR CONVEYS
+THE PROGRAM AS PERMITTED ABOVE, BE LIABLE TO YOU FOR DAMAGES, INCLUDING ANY
+GENERAL, SPECIAL, INCIDENTAL OR CONSEQUENTIAL DAMAGES ARISING OUT OF THE
+USE OR INABILITY TO USE THE PROGRAM (INCLUDING BUT NOT LIMITED TO LOSS OF
+DATA OR DATA BEING RENDERED INACCURATE OR LOSSES SUSTAINED BY YOU OR THIRD
+PARTIES OR A FAILURE OF THE PROGRAM TO OPERATE WITH ANY OTHER PROGRAMS),
+EVEN IF SUCH HOLDER OR OTHER PARTY HAS BEEN ADVISED OF THE POSSIBILITY OF
+SUCH DAMAGES.
+
+  17. Interpretation of Sections 15 and 16.
+
+  If the disclaimer of warranty and limitation of liability provided
+above cannot be given local legal effect according to their terms,
+reviewing courts shall apply local law that most closely approximates
+an absolute waiver of all civil liability in connection with the
+Program, unless a warranty or assumption of liability accompanies a
+copy of the Program in return for a fee.
+
+                     END OF TERMS AND CONDITIONS
+
+            How to Apply These Terms to Your New Programs
+
+  If you develop a new program, and you want it to be of the greatest
+possible use to the public, the best way to achieve this is to make it
+free software which everyone can redistribute and change under these terms.
+
+  To do so, attach the following notices to the program.  It is safest
+to attach them to the start of each source file to most effectively
+state the exclusion of warranty; and each file should have at least
+the "copyright" line and a pointer to where the full notice is found.
+
+    <one line to give the program's name and a brief idea of what it does.>
+    Copyright (C) <year>  <name of author>
+
+    This program is free software: you can redistribute it and/or modify
+    it under the terms of the GNU General Public License as published by
+    the Free Software Foundation, either version 3 of the License, or
+    (at your option) any later version.
+
+    This program is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+    GNU General Public License for more details.
+
+    You should have received a copy of the GNU General Public License
+    along with this program.  If not, see <https://www.gnu.org/licenses/>.
+
+Also add information on how to contact you by electronic and paper mail.
+
+  If the program does terminal interaction, make it output a short
+notice like this when it starts in an interactive mode:
+
+    <program>  Copyright (C) <year>  <name of author>
+    This program comes with ABSOLUTELY NO WARRANTY; for details type `show w'.
+    This is free software, and you are welcome to redistribute it
+    under certain conditions; type `show c' for details.
+
+The hypothetical commands `show w' and `show c' should show the appropriate
+parts of the General Public License.  Of course, your program's commands
+might be different; for a GUI interface, you would use an "about box".
+
+  You should also get your employer (if you work as a programmer) or school,
+if any, to sign a "copyright disclaimer" for the program, if necessary.
+For more information on this, and how to apply and follow the GNU GPL, see
+<https://www.gnu.org/licenses/>.
+
+  The GNU General Public License does not permit incorporating your program
+into proprietary programs.  If your program is a subroutine library, you
+may consider it more useful to permit linking proprietary applications with
+the library.  If this is what you want to do, use the GNU Lesser General
+Public License instead of this License.  But first, please read
+<https://www.gnu.org/licenses/why-not-lgpl.html>.

Makefile

@@ -0,0 +1,33 @@
+BINARY_NAME=subsonic-tui
+BUILD_FOLDER=build
+
+
+.PHONY: build
+build:
+	GOARCH=amd64 GOOS=darwin go build -o ${BUILD_FOLDER}/${BINARY_NAME}-darwin main.go
+	GOARCH=amd64 GOOS=linux go build -o ${BUILD_FOLDER}/${BINARY_NAME}-linux main.go
+	GOARCH=amd64 GOOS=windows go build -o ${BUILD_FOLDER}/${BINARY_NAME}-windows main.go
+
+run: build
+	./${BINARY_NAME}
+
+clean:
+	go clean
+	rm ${BINARY_NAME}-darwin
+	rm ${BINARY_NAME}-linux
+	rm ${BINARY_NAME}-windows
+
+test:
+	go test ./...
+
+test_coverage:
+	go test ./... -coverprofile=coverage.out
+
+dep:
+	go mod download
+
+vet:
+	go vet
+
+lint:
+	golangci-lint run --enable-all

README.md

@@ -0,0 +1,83 @@
+# Subsonic TUI
+
+**NOTE ⚠ !!: This is under heavy development! Do not fork just yet. Im still force pushing to main!**
+
+A Subsonic client and player written in go.
+
+subsonictui (Name in progress) is a simple and easy to use player for Linux, Mac and (yes...) Windows.
+
+<p align="center">
+    <img src="./assets/screenshot.png" alt="Screenshot">
+    <img src="./assets/screenshot-miniplayer.png" alt="Screenshot mini player">
+</p>
+
+## Features
+
+ - [x] Browse Artists/Albums/Playlists
+ - [x] Artist view
+ - [x] Playlist view
+ - [x] Miniplayer on small screen
+ - [x] Album view
+ - [x] Search
+ - [x] Playback
+   - [x] Scrobble (configurable)
+   - [x] Play album
+   - [x] Shuffle album
+   - [x] Add album to queue
+   - [x] Add song to queue
+   - [x] Play Next/Prev
+   - [x] Stop/Pause
+ - [x] Generate artist radio
+ - [x] Generate song radio
+ - [ ] Desktop integration
+ - [ ] Playlist management
+   - [ ] Add song to playlist
+   - [ ] Create playlist from queue
+
+
+## Keybindings
+| Key(s)                                                                       | Action                                 |
+|------------------------------------------------------------------------------|----------------------------------------|
+| <kbd>`</kbd>                                                                 | Focus on Main pane                     |
+| <kbd>1</kbd>                                                                 | Focus on Atrists pane                  |
+| <kbd>2</kbd>                                                                 | Focus on Albums pane                   |
+| <kbd>3</kbd>                                                                 | Focus on Playlists pane                |
+| <kbd>4</kbd>                                                                 | Focus on Queue pane                    |
+| <kbd>Arrow keys</kbd> or <kbd>h</kbd> <kbd>j</kbd> <kbd>k</kbd> <kbd>l</kbd> | Navigation in a pane                   |
+| <kbd>g</kbd>                                                                 | Jump to first item in a list of a pane |
+| <kbd>G</kbd>                                                                 | Jump to last item in a list of a pane  |
+| <kbd>n</kbd>                                                                 | Next song                              |
+| <kbd>N</kbd>                                                                 | Prev song                              |
+| <kbd>q</kbd>                                                                 | Exit                                   |
+| <kbd>r</kbd>                                                                 | While on a song - Start Song radio     |
+| <kbd>s</kbd>                                                                 | Playback - Stop                        |
+| <kbd>p</kbd>                                                                 | Playback - Toggle Play/Pause           |
+| <kbd>c</kbd>                                                                 | Stop, clear queue                      |
+| <kbd>/</kbd>                                                                 | Search                                 |
+| <kbd>?</kbd>                                                                 | Help                                   |
+
+## Config
+`subsonictui` stores a config file at:
+-   Linux: `$HOME/.config/subsonictui/config.yaml`
+-   macOS: `$HOME/Library/Application Support/subsonictui/config.yaml`
+-   Windows: `C:\\Users\%USER%\AppData\Roaming\subsonictui\config.yaml`
+
+
+## Development
+### Build Dependencies
+subsonictui uses [Beep](https://github.com/faiface/beep), that uses [OTO](https://github.com/hajimehoshi/oto) under the hood, so you will need OTO dependencies.
+Mainly on Linux you will need `alsa-devel`.
+
+To Build the project:
+```
+$ make build
+```
+
+## Special Thanks
+subsonictui is built on top of a few projects. I would like to thank them here
+ - [OTO](https://github.com/hajimehoshi/oto)
+ - [Beep](https://github.com/faiface/beep)
+ - [tview](https://github.com/rivo/tview)
+
+Thank you for creating wondeful software that everyone can use. Including myself.
+

go.mod

@@ -0,0 +1,26 @@
+module git.dayanhub.com/sagi/subsonic-tui
+
+go 1.21.5
+
+require (
+	github.com/creasty/defaults v1.7.0
+	github.com/delucks/go-subsonic v0.0.0-20220915164742-2744002c4be5
+	github.com/gdamore/tcell/v2 v2.6.1-0.20231203215052-2917c3801e73
+	github.com/gopxl/beep v1.3.0
+	github.com/rivo/tview v0.0.0-20240116070845-bf8f1c43e46c
+	gopkg.in/yaml.v3 v3.0.1
+)
+
+require (
+	github.com/ebitengine/oto/v3 v3.1.0 // indirect
+	github.com/ebitengine/purego v0.5.0 // indirect
+	github.com/gdamore/encoding v1.0.0 // indirect
+	github.com/hajimehoshi/go-mp3 v0.3.4 // indirect
+	github.com/lucasb-eyer/go-colorful v1.2.0 // indirect
+	github.com/mattn/go-runewidth v0.0.14 // indirect
+	github.com/pkg/errors v0.9.1 // indirect
+	github.com/rivo/uniseg v0.4.3 // indirect
+	golang.org/x/sys v0.12.0 // indirect
+	golang.org/x/term v0.9.0 // indirect
+	golang.org/x/text v0.12.0 // indirect
+)

go.sum

@@ -0,0 +1,77 @@
+github.com/creasty/defaults v1.7.0 h1:eNdqZvc5B509z18lD8yc212CAqJNvfT1Jq6L8WowdBA=
+github.com/creasty/defaults v1.7.0/go.mod h1:iGzKe6pbEHnpMPtfDXZEr0NVxWnPTjb1bbDy08fPzYM=
+github.com/davecgh/go-spew v1.1.1 h1:vj9j/u1bqnvCEfJOwUhtlOARqs3+rkHYY13jYWTU97c=
+github.com/davecgh/go-spew v1.1.1/go.mod h1:J7Y8YcW2NihsgmVo/mv3lAwl/skON4iLHjSsI+c5H38=
+github.com/delucks/go-subsonic v0.0.0-20220915164742-2744002c4be5 h1:RuuxidatioSKGOiBzL1mTY4X22DQD8weEbS3iRLHnAg=
+github.com/delucks/go-subsonic v0.0.0-20220915164742-2744002c4be5/go.mod h1:vnbEuj6Z20PLcHB4rrLQAOXGMjtULfMGhRVSFPcSdUo=
+github.com/ebitengine/oto/v3 v3.1.0 h1:9tChG6rizyeR2w3vsygTTTVVJ9QMMyu00m2yBOCch6U=
+github.com/ebitengine/oto/v3 v3.1.0/go.mod h1:IK1QTnlfZK2GIB6ziyECm433hAdTaPpOsGMLhEyEGTg=
+github.com/ebitengine/purego v0.5.0 h1:JrMGKfRIAM4/QVKaesIIT7m/UVjTj5GYhRSQYwfVdpo=
+github.com/ebitengine/purego v0.5.0/go.mod h1:ah1In8AOtksoNK6yk5z1HTJeUkC1Ez4Wk2idgGslMwQ=
+github.com/gdamore/encoding v1.0.0 h1:+7OoQ1Bc6eTm5niUzBa0Ctsh6JbMW6Ra+YNuAtDBdko=
+github.com/gdamore/encoding v1.0.0/go.mod h1:alR0ol34c49FCSBLjhosxzcPHQbf2trDkoo5dl+VrEg=
+github.com/gdamore/tcell/v2 v2.6.1-0.20231203215052-2917c3801e73 h1:SeDV6ZUSVlTAUUPdMzPXgMyj96z+whQJRRUff8dIeic=
+github.com/gdamore/tcell/v2 v2.6.1-0.20231203215052-2917c3801e73/go.mod h1:pwzJMyH4Hd0AZMJkWQ+/g01dDvYWEvmJuaiRU71Xl8k=
+github.com/gopxl/beep v1.3.0 h1:wlAdb0Ar3q+pPxEspJM5rNFyNGZJ5/pQd6gYh09byU4=
+github.com/gopxl/beep v1.3.0/go.mod h1:gGVz7MJKlfHrmkzr0wSLGNyY7oisM6rFWJnaLjNxEwA=
+github.com/hajimehoshi/go-mp3 v0.3.4 h1:NUP7pBYH8OguP4diaTZ9wJbUbk3tC0KlfzsEpWmYj68=
+github.com/hajimehoshi/go-mp3 v0.3.4/go.mod h1:fRtZraRFcWb0pu7ok0LqyFhCUrPeMsGRSVop0eemFmo=
+github.com/hajimehoshi/oto/v2 v2.3.1/go.mod h1:seWLbgHH7AyUMYKfKYT9pg7PhUu9/SisyJvNTT+ASQo=
+github.com/lucasb-eyer/go-colorful v1.2.0 h1:1nnpGOrhyZZuNyfu1QjKiUICQ74+3FNCN69Aj6K7nkY=
+github.com/lucasb-eyer/go-colorful v1.2.0/go.mod h1:R4dSotOR9KMtayYi1e77YzuveK+i7ruzyGqttikkLy0=
+github.com/mattn/go-runewidth v0.0.14 h1:+xnbZSEeDbOIg5/mE6JF0w6n9duR1l3/WmbinWVwUuU=
+github.com/mattn/go-runewidth v0.0.14/go.mod h1:Jdepj2loyihRzMpdS35Xk/zdY8IAYHsh153qUoGf23w=
+github.com/pkg/errors v0.9.1 h1:FEBLx1zS214owpjy7qsBeixbURkuhQAwrK5UwLGTwt4=
+github.com/pkg/errors v0.9.1/go.mod h1:bwawxfHBFNV+L2hUp1rHADufV3IMtnDRdf1r5NINEl0=
+github.com/pmezard/go-difflib v1.0.0 h1:4DBwDE0NGyQoBHbLQYPwSUPoCMWR5BEzIk/f1lZbAQM=
+github.com/pmezard/go-difflib v1.0.0/go.mod h1:iKH77koFhYxTK1pcRnkKkqfTogsbg7gZNVY4sRDYZ/4=
+github.com/rivo/tview v0.0.0-20240116070845-bf8f1c43e46c h1:l7KIWhypqjpOJzJKynao/m8Ku4Y6PcZDlVcOVirYkKs=
+github.com/rivo/tview v0.0.0-20240116070845-bf8f1c43e46c/go.mod h1:c0SPlNPXkM+/Zgjn/0vD3W0Ds1yxstN7lpquqLDpWCg=
+github.com/rivo/uniseg v0.2.0/go.mod h1:J6wj4VEh+S6ZtnVlnTBMWIodfgj8LQOQFoIToxlJtxc=
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+github.com/stretchr/testify v1.8.4 h1:CcVxjf3Q8PM0mHUKJCdn+eZZtm5yQwehR5yeSVQQcUk=
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+golang.org/x/crypto v0.0.0-20190308221718-c2843e01d9a2/go.mod h1:djNgcEr1/C05ACkg1iLfiJU5Ep61QUkGW8qpdssI0+w=
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+golang.org/x/net v0.6.0/go.mod h1:2Tu9+aMcznHK/AK1HMvgo6xiTLG5rD5rZLDS+rp2Bjs=
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+gopkg.in/check.v1 v0.0.0-20161208181325-20d25e280405 h1:yhCVgyC4o1eVCa2tZl7eS0r+SDo693bJlVdllGtEeKM=
+gopkg.in/check.v1 v0.0.0-20161208181325-20d25e280405/go.mod h1:Co6ibVJAznAaIkqp8huTwlJQCZ016jof/cbN4VW5Yz0=
+gopkg.in/yaml.v3 v3.0.1 h1:fxVm/GzAzEWqLHuvctI91KS9hhNmmWOoWu0XTYJS7CA=
+gopkg.in/yaml.v3 v3.0.1/go.mod h1:K4uyk7z7BCEPqu6E+C64Yfv1cQ7kz7rIZviUmN+EgEM=

internal/client/client.go

@@ -0,0 +1,158 @@
+package client
+
+import (
+	"fmt"
+	"image"
+	"io"
+	"net/http"
+	"sync"
+
+	"git.dayanhub.com/sagi/subsonic-tui/internal/common"
+	"github.com/delucks/go-subsonic"
+)
+
+type Client struct {
+	client subsonic.Client
+}
+
+func NewClient(baseURL string) *Client {
+
+	var client subsonic.Client = subsonic.Client{
+		Client:       &http.Client{},
+		ClientName:   "subsonic-tui",
+		BaseUrl:      baseURL,
+		PasswordAuth: true,
+	}
+
+	return &Client{
+		client: client,
+	}
+
+}
+
+func (c *Client) Authenticate(username, password string) error {
+	c.client.User = username
+	return c.client.Authenticate(password)
+}
+
+func (c *Client) GetUser() (*subsonic.User, error) {
+	return c.client.GetUser(c.client.User)
+}
+
+func (c *Client) GetPlaylists() ([]*subsonic.Playlist, error) {
+	return c.client.GetPlaylists(map[string]string{})
+}
+
+func (c *Client) GetPlaylist(ID string) (*subsonic.Playlist, error) {
+	return c.client.GetPlaylist(ID)
+}
+
+func (c *Client) GetArtists() ([]*subsonic.ArtistID3, error) {
+	indexes, err := c.client.GetArtists(map[string]string{})
+	if err != nil {
+		return nil, err
+	}
+	artists := []*subsonic.ArtistID3{}
+	for _, i := range indexes.Index {
+		artists = append(artists, i.Artist...)
+	}
+
+	return artists, nil
+}
+
+func (c *Client) GetAlbums() ([]*subsonic.AlbumID3, error) {
+	return c.client.GetAlbumList2("alphabeticalByName", map[string]string{
+		"size": "500",
+	})
+	//return c.client.GetAlbumList("*", map[string]string{})
+}
+
+func (c *Client) GetArtist(ID string) (*subsonic.ArtistID3, error) {
+	return c.client.GetArtist(ID)
+}
+
+func (c *Client) GetArtistInfo(ID string) (*subsonic.ArtistInfo2, error) {
+	return c.client.GetArtistInfo2(ID, map[string]string{
+		"count": "20",
+	})
+}
+
+func (c *Client) GetAlbum(ID string) (*subsonic.AlbumID3, error) {
+	return c.client.GetAlbum(ID)
+}
+
+func (c *Client) GetCoverArt(ID string) (image.Image, error) {
+	return c.client.GetCoverArt(ID, map[string]string{
+		"size": "64",
+	})
+}
+
+func (c *Client) GetSimilarSongs(artistID string, maxSongs int) ([]*subsonic.Child, error) {
+	max := fmt.Sprintf("%d", maxSongs)
+	return c.client.GetSimilarSongs2(artistID, map[string]string{
+		"count": max,
+	})
+}
+
+func (c *Client) Stream(ID string) (io.Reader, error) {
+	return c.client.Stream(ID, map[string]string{
+		"format": "mp3",
+	})
+}
+
+func (c *Client) Scrobble(ID string) error {
+	return c.client.Scrobble(ID, map[string]string{})
+}
+
+func (c *Client) GetTopSongs(name string, max int) ([]*subsonic.Child, error) {
+	count := fmt.Sprintf("%d", max)
+	return c.client.GetTopSongs(name, map[string]string{
+		"count": count,
+	})
+}
+
+func (c *Client) Search(query string) (*subsonic.SearchResult3, error) {
+	return c.client.Search3(query, map[string]string{
+		"artistCount": "20",
+		"songCount":   "20",
+		"albumCount":  "20",
+	})
+}
+
+func (c *Client) GetExperimentalArtistRadio(artistId3 *subsonic.ArtistID3, info *subsonic.ArtistInfo2, max int) ([]*subsonic.Child, error) {
+	var wg sync.WaitGroup
+	ID := artistId3.ID
+	similarArtists := info.SimilarArtist
+	songs := []*subsonic.Child{}
+	similarArtistsSongs := 10
+	thisArtistFactor := 3
+	portion := len(info.SimilarArtist) * similarArtistsSongs * thisArtistFactor
+	wg.Add(2)
+	go func() {
+		s, _ := c.GetSimilarSongs(ID, portion)
+		songs = append(songs, s...)
+		wg.Done()
+	}()
+	go func() {
+		s, _ := c.GetTopSongs(artistId3.Name, similarArtistsSongs)
+		songs = append(songs, s...)
+		wg.Done()
+	}()
+	common.ShuffleSlice(similarArtists)
+	for _, a := range similarArtists {
+		wg.Add(1)
+		artist := a
+		go func() {
+			s, _ := c.GetSimilarSongs(artist.ID, similarArtistsSongs)
+			songs = append(songs, s...)
+			wg.Done()
+		}()
+	}
+	wg.Wait()
+	if max > len(songs) {
+		max = len(songs)
+	}
+	songs = songs[:max]
+	common.ShuffleSlice(songs)
+	return songs, nil
+}

internal/common/shuffle.go

@@ -0,0 +1,16 @@
+package common
+
+import (
+	"math/rand"
+	"reflect"
+)
+
+func ShuffleSlice(slice interface{}) {
+	rv := reflect.ValueOf(slice)
+	swap := reflect.Swapper(slice)
+	length := rv.Len()
+	for i := length - 1; i > 0; i-- {
+		j := rand.Intn(i + 1)
+		swap(i, j)
+	}
+}

internal/config/config.go

@@ -0,0 +1,124 @@
+package config
+
+import (
+	"encoding/base64"
+	"fmt"
+	"os"
+	"path"
+	"strings"
+
+	"gopkg.in/yaml.v3"
+
+	"github.com/creasty/defaults"
+)
+
+var configPath string
+
+type _config struct {
+	Username              string `yaml:"username"`
+	Password              string `yaml:"password"`
+	URL                   string `yaml:"url"`
+	EnableScrobble        bool   `yaml:"enable_scrobble" default:"false"`
+	MaxRadioSongs         int    `yaml:"max_radio_songs" default:"50"`
+	ExperimentalRadioAlgo bool   `yaml:"experimental_radio_algo" default:"false"`
+}
+
+var configStruct *_config
+
+func init() {
+	userConfigDir, err := os.UserConfigDir()
+	configDir := path.Join(userConfigDir, "subsonictui")
+	configPath = path.Join(configDir, "config.yaml")
+
+	if err != nil {
+		fmt.Printf("[ERROR] Failed to fetch user config directory. %e\n", err)
+		os.Exit(1)
+	}
+	if _, err := os.Stat(configDir); os.IsNotExist(err) {
+		err := os.MkdirAll(configDir, 0700)
+		if err != nil {
+			panic(err)
+		}
+	}
+	var configFile *os.File
+	if _, err := os.Stat(configPath); os.IsNotExist(err) {
+		configFile, err = os.Create(configPath)
+		defer func() {
+			err := configFile.Close()
+			if err != nil {
+				panic(err)
+			}
+		}()
+		if err != nil {
+			fmt.Printf("[ERROR] Failed to create config file @ %s. %e\n", configPath, err)
+			os.Exit(1)
+		}
+	}
+	configStruct, err = loadConfig()
+	if err != nil {
+		fmt.Printf("[ERROR] Failed to load config file @ %s. %e\n", configPath, err)
+		os.Exit(1)
+	}
+	fmt.Printf("Init Config %s\n", configPath)
+}
+
+func URL() string {
+	return configStruct.URL
+}
+func Username() string {
+	return configStruct.Username
+}
+func Password() string {
+	p, _ := base64.StdEncoding.DecodeString(configStruct.Password)
+	return strings.TrimSpace(string(p))
+}
+func ScrobbleEnabled() bool {
+	return configStruct.EnableScrobble
+}
+func MaxRadioSongs() int {
+	return configStruct.MaxRadioSongs
+}
+func ExperimentalRadioAlgo() bool {
+	return configStruct.ExperimentalRadioAlgo
+}
+
+func SetPassword(p string) {
+	configStruct.Password = base64.StdEncoding.EncodeToString([]byte(p))
+}
+
+func SetUsername(u string) {
+	configStruct.Username = u
+}
+
+func SetURL(u string) {
+	configStruct.URL = u
+}
+
+func loadConfig() (*_config, error) {
+	c := &_config{}
+	err := defaults.Set(c)
+	if err != nil {
+		panic(err)
+	}
+	file, err := os.ReadFile(configPath)
+	if err != nil {
+		return nil, err
+	}
+	if err = yaml.Unmarshal(file, c); err != nil {
+		return nil, err
+	}
+	return c, nil
+}
+
+func SaveConfig() {
+	yml, err := yaml.Marshal(configStruct)
+	if err != nil {
+		fmt.Printf("[ERROR] Failed to convert config to yaml. %e\n", err)
+		os.Exit(1)
+	}
+	err = os.WriteFile(configPath, yml, 0644)
+	if err != nil {
+		fmt.Printf("[ERROR] Failed to save config file @ %s. %e\n", configPath, err)
+		os.Exit(1)
+	}
+}

internal/playback/controller.go

@@ -0,0 +1,149 @@
+package playback
+
+import (
+	"io"
+	"sync"
+	"time"
+
+	"github.com/delucks/go-subsonic"
+	"github.com/gopxl/beep"
+	"github.com/gopxl/beep/mp3"
+	"github.com/gopxl/beep/speaker"
+)
+
+type Controller struct {
+	stream          beep.StreamSeekCloser
+	song            *subsonic.Child
+	songElapsedFunc func(elapsed time.Duration)
+	format          *beep.Format
+	closeChan       chan bool
+	songEndedFunc   func(song *subsonic.Child)
+	isPlaying       bool
+	ctrl            *beep.Ctrl
+	tickerRunning   bool
+	controllerLock  *sync.Mutex
+}
+
+func NewController() *Controller {
+	controller := &Controller{
+		closeChan:      make(chan bool),
+		controllerLock: &sync.Mutex{},
+	}
+	return controller
+}
+
+func (c *Controller) Playing() bool {
+	return c.isPlaying
+}
+
+func (c *Controller) SetSongElapsedFunc(f func(elapsed time.Duration)) {
+	c.songElapsedFunc = f
+}
+
+func (c *Controller) SetSongEndedFunc(f func(song *subsonic.Child)) {
+	c.songEndedFunc = f
+}
+
+func (c *Controller) Close() error {
+	if c.tickerRunning {
+		c.closeChan <- true
+	}
+	if c.stream != nil {
+		c.Stop()
+		speaker.Close()
+	}
+	return nil
+}
+
+func (c *Controller) TogglePlayPause() {
+	if c.ctrl == nil {
+		if c.stream != nil {
+			c.ctrl.Streamer = c.stream
+			c.ctrl.Paused = false
+			return
+		}
+		return
+	}
+	c.ctrl.Paused = !c.ctrl.Paused
+}
+
+func (c *Controller) Stop() {
+	if c.ctrl == nil {
+		return
+	}
+	go func() {
+		c.controllerLock.Lock()
+		speaker.Clear()
+		c.ctrl.Paused = true
+		c.isPlaying = false
+		c.ctrl = nil
+		c.stream = nil
+		c.song = nil
+		c.songElapsedFunc(time.Duration(0))
+		c.controllerLock.Unlock()
+	}()
+}
+
+func (c *Controller) playbackTicker(format beep.Format) {
+	if c.tickerRunning {
+		return
+	}
+	c.tickerRunning = true
+	for {
+		c.controllerLock.Lock()
+		select {
+		case <-c.closeChan:
+			c.tickerRunning = false
+			c.controllerLock.Unlock()
+			return
+		default:
+			if c.isPlaying && c.stream != nil {
+				pos := c.stream.Position()
+				songDuration := time.Duration(c.song.Duration) * time.Second
+				elapsed := format.SampleRate.D(pos).Round(time.Second)
+				if elapsed >= songDuration {
+					elapsed = songDuration
+					c.songElapsedFunc(elapsed)
+					c.tickerRunning = false
+					c.controllerLock.Unlock()
+					c.songEndedFunc(c.song)
+					return
+				}
+				c.songElapsedFunc(elapsed)
+			}
+		}
+		c.controllerLock.Unlock()
+		time.Sleep(time.Second)
+	}
+}
+
+func (c *Controller) Stream(reader io.Reader, song *subsonic.Child) {
+	c.Stop()
+
+	readerCloser := io.NopCloser(reader)
+	decodedMp3, format, err := mp3.Decode(readerCloser)
+	decodedMp3.Position()
+	if err != nil {
+		panic("mp3.NewDecoder failed: " + err.Error())
+	}
+
+	if c.format == nil {
+		c.format = &format
+	}
+
+	var stream beep.Streamer = decodedMp3
+	if err = speaker.Init(format.SampleRate, format.SampleRate.N(time.Second/10)); err != nil {
+		stream = beep.Resample(3, format.SampleRate, c.format.SampleRate, decodedMp3)
+	}
+
+	c.stream = decodedMp3
+	c.song = song
+
+	ctrl := &beep.Ctrl{Streamer: stream}
+	speaker.Play(beep.Seq(ctrl))
+	c.ctrl = ctrl
+	c.isPlaying = true
+	if !c.tickerRunning {
+		go c.playbackTicker(format)
+	}
+}

internal/tui/tui.go

@@ -0,0 +1,124 @@
+package tui
+
+import (
+	"fmt"
+	"os"
+
+	"git.dayanhub.com/sagi/subsonic-tui/internal/client"
+	"git.dayanhub.com/sagi/subsonic-tui/internal/config"
+	"git.dayanhub.com/sagi/subsonic-tui/internal/playback"
+	"git.dayanhub.com/sagi/subsonic-tui/internal/tui/views"
+	"github.com/gdamore/tcell/v2"
+	"github.com/rivo/tview"
+)
+
+type TUI struct {
+	app    *tview.Application
+	layout views.View
+
+	client      *client.Client
+	playbackCtl *playback.Controller
+}
+
+func NewLogin() *TUI {
+	app := tview.NewApplication()
+	layout := views.NewLoginView(func(u, p, url string) {
+		c := client.NewClient(url)
+		err := c.Authenticate(u, p)
+		if err != nil {
+			app.Stop()
+			fmt.Printf("[Error] Failed to login. Aborting %e", err)
+			os.Exit(1)
+		}
+		config.SetURL(url)
+		config.SetUsername(u)
+		config.SetPassword(p)
+		config.SaveConfig()
+		app.Stop()
+	}, func() {
+		app.Stop()
+		os.Exit(0)
+	})
+	app.EnableMouse(true).
+		SetRoot(layout.GetView(), true)
+
+	app.SetInputCapture(func(event *tcell.EventKey) *tcell.EventKey {
+		if event.Key() == tcell.KeyEsc {
+			app.Stop()
+			os.Exit(0)
+		}
+		return event
+	})
+
+	return &TUI{
+		app:    app,
+		layout: layout,
+	}
+}
+
+func NewPlayer(client *client.Client, playbackCtl *playback.Controller) *TUI {
+	app := tview.NewApplication()
+
+	layout := views.NewLayout(client, playbackCtl, func() {
+		go app.Draw()
+	})
+	help := views.NewHelp()
+
+	pages := tview.NewPages()
+	pages.AddPage("app", layout.GetView(), true, true)
+	pages.AddPage("help", help.GetView(), true, false)
+
+	help.SetKeyPressedFunc(func() {
+		pages.SwitchToPage("app")
+	})
+	app.EnableMouse(true).
+		SetRoot(pages, true)
+
+	app.SetInputCapture(func(event *tcell.EventKey) *tcell.EventKey {
+		if currentPage, _ := pages.GetFrontPage(); currentPage == "help" {
+			pages.SwitchToPage("app")
+			help.GetView().Blur()
+			layout.GetView().Focus(nil)
+			return nil
+		}
+		if layout.Mode() == views.StatusModeSearch {
+			return event
+		}
+		if event.Rune() == 'q' {
+			app.Stop()
+			fmt.Println("Exiting..")
+			return nil
+		} else if event.Rune() == 'h' {
+			return tcell.NewEventKey(tcell.KeyLeft, rune(tcell.KeyLeft), event.Modifiers())
+		} else if event.Rune() == 'j' {
+			return tcell.NewEventKey(tcell.KeyDown, rune(tcell.KeyDown), event.Modifiers())
+		} else if event.Rune() == 'k' {
+			return tcell.NewEventKey(tcell.KeyUp, rune(tcell.KeyUp), event.Modifiers())
+		} else if event.Rune() == 'l' {
+			return tcell.NewEventKey(tcell.KeyRight, rune(tcell.KeyRight), event.Modifiers())
+		} else if event.Rune() == '?' {
+			pages.SwitchToPage("help")
+			layout.GetView().Blur()
+			go app.Draw()
+		}
+		return event
+	})
+	app.SetAfterDrawFunc(func(screen tcell.Screen) {
+		layout.Update()
+	})
+	app.GetFocus().Blur()
+
+	//app.SetFocus(layout.GetView())
+
+	return &TUI{
+		app:         app,
+		layout:      layout,
+		client:      client,
+		playbackCtl: playbackCtl,
+	}
+
+}
+
+func (t *TUI) Run() error {
+	return t.app.Run()
+}

internal/tui/views/albums.go

@@ -0,0 +1,73 @@
+package views
+
+import (
+	"git.dayanhub.com/sagi/subsonic-tui/internal/client"
+	"github.com/delucks/go-subsonic"
+	"github.com/gdamore/tcell/v2"
+	"github.com/rivo/tview"
+)
+
+var _ View = &albums{}
+
+type albums struct {
+	view     *tview.Table
+	client   *client.Client
+	albums   []*subsonic.AlbumID3
+	callback func(albumID string)
+}
+
+func NewAlbums(client *client.Client) *albums {
+
+	list := tview.NewTable()
+
+	list.SetBackgroundColor(tcell.ColorBlack)
+	list.SetTitle("Albums [2]")
+	list.SetBorder(true)
+
+	resp, _ := client.GetAlbums()
+
+	obj := &albums{
+		view:   list,
+		client: client,
+		albums: resp,
+	}
+
+	list.SetSelectedFunc(func(row, column int) {
+		obj.callback(obj.albums[row].ID)
+	})
+
+	list.SetFocusFunc(func() {
+		list.SetBorderColor(tcell.ColorRed)
+		list.SetSelectable(true, false)
+	})
+	list.SetBlurFunc(func() {
+		list.SetBorderColor(tcell.ColorWhite)
+		list.SetSelectable(false, false)
+	})
+
+	obj.Update()
+	return obj
+}
+
+func (a *albums) SetAlbums(al []*subsonic.AlbumID3) {
+	a.albums = al
+	a.Update()
+}
+
+func (a *albums) Update() {
+	a.view.Clear()
+	for i, pl := range a.albums {
+		title := tview.NewTableCell(pl.Name).SetExpansion(1).SetMaxWidth(15)
+		artist := tview.NewTableCell(pl.Artist).SetExpansion(1).SetAlign(tview.AlignRight)
+		a.view.SetCell(i, 0, title)
+		a.view.SetCell(i, 1, artist)
+	}
+}
+
+func (a *albums) SetCallback(f func(albumID string)) {
+	a.callback = f
+}
+
+func (a *albums) GetView() tview.Primitive {
+	return a.view
+}

internal/tui/views/artists.go

@@ -0,0 +1,70 @@
+package views
+
+import (
+	"git.dayanhub.com/sagi/subsonic-tui/internal/client"
+	"github.com/delucks/go-subsonic"
+	"github.com/gdamore/tcell/v2"
+	"github.com/rivo/tview"
+)
+
+var _ View = &artists{}
+
+type artists struct {
+	view             *tview.Table
+	client           *client.Client
+	artists          []*subsonic.ArtistID3
+	selectArtistFunc func(artistId string)
+	openArtistFunc   func(artistId string)
+}
+
+func NewArtists(client *client.Client) *artists {
+
+	list := tview.NewTable()
+
+	list.SetBackgroundColor(tcell.ColorBlack)
+	list.SetTitle("Artists [1]")
+	list.SetBorder(true)
+	list.SetFocusFunc(func() {
+		list.SetBorderColor(tcell.ColorRed)
+		list.SetSelectable(true, false)
+	})
+	list.SetBlurFunc(func() {
+		list.SetBorderColor(tcell.ColorWhite)
+		list.SetSelectable(false, false)
+	})
+
+	arts, _ := client.GetArtists()
+
+	for i, artist := range arts {
+		cell := tview.NewTableCell(artist.Name).SetExpansion(1)
+		list.SetCell(i, 0, cell)
+		//list.AddItem(artist.Name, fmt.Sprintf("%s", artist.Name), '0', nil)
+	}
+
+	resp := &artists{
+		view:    list,
+		client:  client,
+		artists: arts,
+	}
+
+	list.SetSelectedFunc(func(row, column int) {
+		resp.openArtistFunc(resp.artists[row].ID)
+	})
+
+	return resp
+}
+
+func (a *artists) SetSelectArtistFunc(f func(artistId string)) {
+	a.selectArtistFunc = f
+}
+func (a *artists) SetOpenArtistFunc(f func(artistId string)) {
+	a.openArtistFunc = f
+}
+
+func (a *artists) Update() {
+
+}
+
+func (a *artists) GetView() tview.Primitive {
+	return a.view
+}

internal/tui/views/help.go

@@ -0,0 +1,109 @@
+package views
+
+import (
+	"github.com/gdamore/tcell/v2"
+	"github.com/rivo/tview"
+)
+
+var _ View = &help{}
+
+type help struct {
+	view           *tview.Flex
+	keyPressedFunc func()
+}
+
+type keyMap struct {
+	key         string
+	description string
+}
+
+var keyMaps = []keyMap{
+	{
+		"Arrows", "Navigation",
+	},
+	{
+		"h, j, k, l", "Navigation",
+	},
+	{
+		"g", "Jump to top",
+	},
+	{
+		"G", "Jump to bottom",
+	},
+	{
+		"p", "Toggle Play/Pause",
+	},
+	{
+		"s", "Stop",
+	},
+	{
+		"n", "Play next song",
+	},
+	{
+		"N", "Play previous song",
+	},
+	{
+		"c", "Stop and clear queue",
+	},
+	{
+		"r", "Start song radio",
+	},
+	{
+		"/", "Search",
+	},
+	{
+		"q", "Quit",
+	},
+}
+
+func NewHelp() *help {
+	h := &help{}
+	view := tview.NewTable()
+	height := 16
+	width := 100
+	view.SetBorder(true)
+	view.SetTitle("  Keybindings  ")
+	view.SetTitleAlign(tview.AlignCenter)
+
+	for i, km := range keyMaps {
+		odd := i%2 != 0
+		txtColor := tcell.ColorWhite
+		if odd {
+			txtColor = tcell.ColorYellow
+		}
+		keyCell := tview.NewTableCell(km.key).
+			SetExpansion(1).
+			SetAlign(tview.AlignCenter).
+			SetTextColor(txtColor)
+		descCell := tview.NewTableCell(km.description).
+			SetExpansion(1).
+			SetAlign(tview.AlignCenter).
+			SetTextColor(txtColor)
+		view.SetCell(i, 0, keyCell)
+		view.SetCell(i, 1, descCell)
+	}
+
+	wrapper := tview.NewFlex().
+		AddItem(nil, 0, 1, false).
+		AddItem(tview.NewFlex().SetDirection(tview.FlexRow).
+			AddItem(nil, 0, 1, false).
+			AddItem(view, height, 1, true).
+			AddItem(nil, 0, 1, false), width, 1, true).
+		AddItem(nil, 0, 1, false)
+
+	h.view = wrapper
+
+	return h
+}
+
+func (h *help) GetView() tview.Primitive {
+	return h.view
+}
+
+func (h *help) Update() {
+
+}
+
+func (h *help) SetKeyPressedFunc(f func()) {
+	h.keyPressedFunc = f
+}

internal/tui/views/layout.go

@@ -0,0 +1,318 @@
+package views
+
+import (
+	"time"
+
+	"git.dayanhub.com/sagi/subsonic-tui/internal/client"
+	"git.dayanhub.com/sagi/subsonic-tui/internal/config"
+	"git.dayanhub.com/sagi/subsonic-tui/internal/playback"
+	"github.com/delucks/go-subsonic"
+	"github.com/gdamore/tcell/v2"
+	"github.com/rivo/tview"
+)
+
+var _ View = &layout{}
+
+type layout struct {
+	view               *tview.Pages
+	player             View
+	currentFocusedView tview.Primitive
+	smallView          *tview.Flex
+	largeView          *tview.Grid
+	status             *statusLine
+	mainView           *main
+}
+
+func NewLayout(client *client.Client, playbackCtl *playback.Controller, refreshUI func()) *layout {
+	layout := &layout{}
+
+	largeView := tview.NewGrid().SetRows(0, 4, 1)
+	smallView := tview.NewFlex().SetDirection(tview.FlexRow)
+	layout.largeView = largeView
+	layout.smallView = smallView
+	pages := tview.NewPages()
+
+	// Status / command / search
+	statusLine := NewStatusLine()
+	layout.status = statusLine
+
+	// Side Panel (Artist/Albums/Playlist)
+	albums := NewAlbums(client)
+	artists := NewArtists(client)
+	playlists := NewPlaylists(client)
+
+	sidePanel := tview.NewGrid().SetRows(0, 0, 0)
+	sidePanel.AddItem(artists.GetView(), 0, 0, 1, 1, 0, 0, false)
+	sidePanel.AddItem(albums.GetView(), 1, 0, 1, 1, 0, 0, false)
+	sidePanel.AddItem(playlists.GetView(), 2, 0, 1, 1, 0, 0, false)
+
+	// main pane
+	main := NewMainView(client, statusLine.Log)
+	layout.mainView = main
+	// Queue
+	queue := NewQueue()
+
+	// Main view
+	mainView := tview.NewFlex()
+	mainView.AddItem(sidePanel, 0, 1, false)
+	mainView.AddItem(main.GetView(), 0, 2, false)
+	mainView.AddItem(queue.GetView(), 0, 1, false)
+	largeView.AddItem(mainView, 0, 0, 1, 3, 0, 0, false)
+	// Player
+	player := NewPlayer(client)
+	largeView.AddItem(player.GetView(), 1, 0, 1, 3, 0, 0, false)
+
+	// Add status line
+	largeView.AddItem(statusLine.GetView(), 2, 0, 1, 3, 0, 0, false)
+
+	// Callbacks
+	artists.SetSelectArtistFunc(func(artistId string) {
+		artists.view.Blur()
+		statusLine.Log("Fetching artist's albums...")
+		go func() {
+			a, _ := client.GetArtist(artistId)
+			albums.SetAlbums(a.Album)
+			albums.GetView().Focus(nil)
+			refreshUI()
+		}()
+	})
+	artists.SetOpenArtistFunc(func(artistId string) {
+		artists.view.Blur()
+		statusLine.Log("Fetching Artists...")
+		layout.currentFocusedView = main.GetView()
+		layout.rebuildSmallView()
+		go func() {
+			a, _ := client.GetArtist(artistId)
+			main.SetArtist(a)
+			main.GetView().Focus(nil)
+			refreshUI()
+		}()
+	})
+
+	albums.SetCallback(func(albumID string) {
+		albums.view.Blur()
+		statusLine.Log("Fetching album...")
+		layout.currentFocusedView = main.GetView()
+		layout.rebuildSmallView()
+		go func() {
+			a, _ := client.GetAlbum(albumID)
+			main.SetAlbum(a)
+			main.view.Focus(nil)
+			refreshUI()
+		}()
+	})
+
+	playlists.SetCallback(func(p *subsonic.Playlist) {
+		playlists.view.Blur()
+		statusLine.Log("Fetching playlist...")
+		layout.currentFocusedView = main.GetView()
+		layout.rebuildSmallView()
+		go func() {
+			playlist, _ := client.GetPlaylist(p.ID)
+			main.SetPlaylist(playlist)
+			main.view.Focus(nil)
+			refreshUI()
+		}()
+	})
+
+	main.SetPlayAllFunc(func(songs ...*subsonic.Child) {
+		statusLine.Log("Loaded #%d songs.", len(songs))
+		queue.ReplaceAll(songs...)
+	})
+	main.SetPlayAddSongFunc(func(song ...*subsonic.Child) {
+		statusLine.Log("Added #%d songs to queue.", len(song))
+		queue.Add(song...)
+	})
+
+	playbackCtl.SetSongElapsedFunc(func(elapsed time.Duration) {
+		player.UpdateProgress(elapsed)
+		queue.Update()
+		refreshUI()
+	})
+
+	playbackCtl.SetSongEndedFunc(func(song *subsonic.Child) {
+		statusLine.Log("")
+		queue.Next()
+		if config.ScrobbleEnabled() {
+			// Scrobble
+			_ = client.Scrobble(song.ID)
+		}
+	})
+
+	queue.SetPlayFunc(func(song *subsonic.Child) {
+		reader, _ := client.Stream(song.ID)
+
+		playbackCtl.Stream(reader, song)
+		player.SetSongInfo(song)
+	})
+
+	statusLine.SetOnUpdateFunc(func() {
+		refreshUI()
+	})
+
+	statusLine.SetSearchFunc(func(quary string) {
+		if len(quary) == 0 {
+			layout.currentFocusedView.Focus(nil)
+			statusLine.Log("Search canceled")
+			return
+		}
+		// Search...
+		statusLine.Log("Searching for '%s'....", quary)
+		statusLine.view.Blur()
+		go func() {
+			result, _ := client.Search(quary)
+			layout.currentFocusedView.Blur()
+			main.SetSearch(result, quary)
+			layout.currentFocusedView = main.GetView()
+			layout.rebuildSmallView()
+			refreshUI()
+		}()
+	})
+
+	// Key Bindings
+	pages.SetInputCapture(func(event *tcell.EventKey) *tcell.EventKey {
+		if statusLine.Mode() == StatusModeSearch {
+			return event
+		}
+		if event.Rune() == '1' {
+			// Focus Artists
+			artists.view.Blur()
+			albums.view.Blur()
+			playlists.view.Blur()
+			main.view.Blur()
+			queue.view.Blur()
+			artists.view.Focus(nil)
+			layout.currentFocusedView = artists.GetView()
+			layout.rebuildSmallView()
+			return nil
+		} else if event.Rune() == '2' {
+			// Focus Albums
+			artists.view.Blur()
+			albums.view.Blur()
+			playlists.view.Blur()
+			main.view.Blur()
+			queue.view.Blur()
+			albums.view.Focus(nil)
+			layout.currentFocusedView = albums.GetView()
+			layout.rebuildSmallView()
+			return nil
+		} else if event.Rune() == '3' {
+			// Focus Playlists
+			artists.view.Blur()
+			albums.view.Blur()
+			playlists.view.Blur()
+			main.view.Blur()
+			queue.view.Blur()
+			playlists.view.Focus(nil)
+			layout.currentFocusedView = playlists.GetView()
+			layout.rebuildSmallView()
+			return nil
+		} else if event.Rune() == '`' {
+			// Focus Songs
+			artists.view.Blur()
+			albums.view.Blur()
+			playlists.view.Blur()
+			main.view.Blur()
+			queue.view.Blur()
+			main.view.Focus(nil)
+			layout.currentFocusedView = main.GetView()
+			layout.rebuildSmallView()
+			return nil
+		} else if event.Rune() == '4' {
+			// Focus Queue
+			artists.view.Blur()
+			albums.view.Blur()
+			playlists.view.Blur()
+			main.view.Blur()
+			queue.view.Blur()
+			queue.view.Focus(nil)
+			layout.currentFocusedView = queue.GetView()
+			layout.rebuildSmallView()
+			return nil
+		} else if event.Rune() == 'n' {
+			queue.Next()
+			return nil
+
+		} else if event.Rune() == 'N' {
+			queue.Prev()
+			return nil
+		} else if event.Rune() == 's' {
+			playbackCtl.Stop()
+			return nil
+		} else if event.Rune() == 'p' {
+			if !playbackCtl.Playing() {
+				queue.PlayCurrent()
+				return nil
+			}
+			playbackCtl.TogglePlayPause()
+			return nil
+		} else if event.Rune() == 'c' {
+			playbackCtl.Stop()
+			queue.Clear()
+			return nil
+		} else if event.Rune() == '/' {
+			layout.currentFocusedView.Blur()
+			statusLine.Search()
+			refreshUI()
+			return nil
+		}
+
+		return event
+	})
+
+	largeView.SetFocusFunc(func() {
+		largeView.Blur()
+		layout.currentFocusedView.Focus(nil)
+	})
+	smallView.SetFocusFunc(func() {
+		smallView.Blur()
+		layout.currentFocusedView.Focus(nil)
+	})
+
+	statusLine.Log("Press '?' for help")
+
+	pages.AddPage("small", smallView, true, false)
+	pages.AddPage("large", largeView, true, true)
+
+	layout.view = pages
+	layout.player = player
+	//Auto focus on artists
+	artists.GetView().Focus(nil)
+	layout.currentFocusedView = artists.GetView()
+	layout.rebuildSmallView()
+	return layout
+}
+func (l *layout) rebuildSmallView() {
+	l.smallView.Clear()
+	l.smallView.AddItem(l.currentFocusedView, 0, 1, false)
+	l.smallView.AddItem(l.player.GetView(), 4, 0, false)
+	l.smallView.AddItem(l.status.GetView(), 1, 0, false)
+}
+
+func (l *layout) Mode() Statusmode {
+	return l.status.Mode()
+}
+
+func (l *layout) GetView() tview.Primitive {
+	return l.view
+}
+
+func (l *layout) Update() {
+	_, _, w, h := l.view.GetRect()
+	page, _ := l.view.GetFrontPage()
+	smallView := w < 100 || h < 30
+	if smallView {
+		if page != "small" {
+			l.mainView.SetMiniView(true)
+			go l.view.SwitchToPage("small")
+			return
+		}
+	} else {
+		if page != "large" {
+			l.mainView.SetMiniView(false)
+			go l.view.SwitchToPage("large")
+			return
+		}
+	}
+	l.player.Update()
+}

internal/tui/views/login.go

@@ -0,0 +1,73 @@
+package views
+
+import (
+	"github.com/rivo/tview"
+)
+
+var _ View = &login{}
+
+const (
+	ViewEventLoginClicked = "login-clicked"
+)
+
+type login struct {
+	view      *tview.Flex
+	form      *tview.Form
+	username  string
+	password  string
+	url       string
+	loginFunc func(u, p, url string)
+	exitFunc  func()
+}
+
+func (l *login) GetView() tview.Primitive {
+	return l.view
+}
+
+func (l *login) onFormLogin() {
+	l.loginFunc(l.username, l.password, l.url)
+}
+
+func NewLoginView(loginFunc func(u, p, url string), exitFunc func()) View {
+	l := &login{}
+	l.loginFunc = loginFunc
+	l.exitFunc = exitFunc
+
+	form := tview.NewForm().
+		AddInputField("Server URL", "https://", 20, nil, func(text string) {
+			l.url = text
+		}).
+		AddInputField("Username", "", 20, nil, func(text string) {
+			l.username = text
+		}).
+		AddPasswordField("Password", "", 20, '*', func(text string) {
+			l.password = text
+		}).
+		AddButton("Login", l.onFormLogin).
+		AddButton("Quit", func() {
+			l.exitFunc()
+		})
+	form.SetBorder(true).SetTitle(" Login ").SetTitleAlign(tview.AlignCenter)
+
+	width := 50
+	height := 20
+
+	wrapper := tview.NewFlex().
+		AddItem(nil, 0, 1, false).
+		AddItem(tview.NewFlex().SetDirection(tview.FlexRow).
+			AddItem(nil, 0, 1, false).
+			AddItem(form, height, 1, true).
+			AddItem(nil, 0, 1, false), width, 1, true).
+		AddItem(nil, 0, 1, false)
+
+	l.view = wrapper
+	l.form = form
+
+	l.GetView().Focus(func(p tview.Primitive) {})
+	form.SetFocus(0)
+	return l
+}
+
+func (l *login) Update() {
+
+}

internal/tui/views/main.go

@@ -0,0 +1,608 @@
+package views
+
+import (
+	"fmt"
+	"strconv"
+	"time"
+
+	"git.dayanhub.com/sagi/subsonic-tui/internal/client"
+	"git.dayanhub.com/sagi/subsonic-tui/internal/common"
+	"git.dayanhub.com/sagi/subsonic-tui/internal/config"
+	"github.com/delucks/go-subsonic"
+	"github.com/gdamore/tcell/v2"
+	"github.com/rivo/tview"
+)
+
+var _ View = &main{}
+
+type mainviewmode int
+
+const (
+	mainModeAlbum    mainviewmode = iota
+	mainModePlaylist mainviewmode = iota
+	mainModeArtist   mainviewmode = iota
+	mainModeSearch   mainviewmode = iota
+)
+
+type main struct {
+	view         *tview.Flex
+	client       *client.Client
+	mode         mainviewmode
+	query        string
+	miniView     bool
+	album        *subsonic.AlbumID3
+	searchResult *subsonic.SearchResult3
+	playlist     *subsonic.Playlist
+	songList     *tview.Table
+	artist       *subsonic.ArtistID3
+	artistInfo   *subsonic.ArtistInfo2
+	playAllFunc  func(song ...*subsonic.Child)
+	addSongsFunc func(song ...*subsonic.Child)
+	log          func(format string, a ...any)
+}
+
+func NewMainView(client *client.Client, log func(format string, a ...any)) *main {
+	playlistAlbum := &main{
+		client: client,
+		log:    log,
+	}
+
+	flex := tview.NewFlex().SetDirection(tview.FlexRow)
+
+	flex.SetBackgroundColor(tcell.ColorBlack)
+	flex.SetTitle("Subsonic TUI [`]")
+	flex.SetBorder(true)
+	flex.SetFocusFunc(func() {
+		flex.SetBorderColor(tcell.ColorRed)
+		if playlistAlbum.songList != nil {
+			//playlistAlbum.view.Blur()
+			playlistAlbum.songList.Focus(nil)
+		}
+	})
+	flex.SetBlurFunc(func() {
+		flex.SetBorderColor(tcell.ColorWhite)
+		if playlistAlbum.songList != nil {
+			playlistAlbum.songList.Blur()
+		}
+	})
+
+	playlistAlbum.view = flex
+	// Empty Box for starters...
+	playlistAlbum.view.AddItem(tview.NewBox(), 0, 1, false)
+	return playlistAlbum
+}
+
+func (m *main) SetMiniView(mini bool) {
+	m.miniView = mini
+}
+
+func (m *main) SetAlbum(album *subsonic.AlbumID3) {
+	m.mode = mainModeAlbum
+	m.album = album
+	m.Update()
+	m.log("")
+}
+
+func (m *main) SetPlaylist(playlist *subsonic.Playlist) {
+	m.mode = mainModePlaylist
+	m.playlist = playlist
+	m.Update()
+	m.log("")
+}
+
+func (m *main) SetArtist(artist *subsonic.ArtistID3) {
+	m.mode = mainModeArtist
+	m.artist = artist
+	info, _ := m.client.GetArtistInfo(artist.ID)
+	m.artistInfo = info
+	m.Update()
+	m.log("")
+}
+
+func (m *main) SetSearch(result *subsonic.SearchResult3, query string) {
+	m.mode = mainModeSearch
+	m.searchResult = result
+	m.query = query
+	m.Update()
+	m.log("Found #%d artists, #%d albums and #%d songs",
+		len(result.Artist),
+		len(result.Album),
+		len(result.Song),
+	)
+}
+
+func (m *main) drawPlaylist() {
+	subtitle := fmt.Sprintf("%s\n\nCreated by: %s | %s", m.playlist.Comment, m.playlist.Owner, time.Duration(m.playlist.Duration*int(time.Second)).String())
+	m.populateHeader(m.playlist.Name, subtitle, m.playlist.Duration, m.playlist.CoverArt)
+	playBtn := m.drawPlaylistAlbumButtons(m.playlist.Entry)
+	m.populateSongs(m.playlist.Entry, playBtn)
+}
+
+func (m *main) drawAlbum() {
+	subtitle := fmt.Sprintf("%s\n\n%d | %s", m.album.Artist, m.album.Year, time.Duration(m.album.Duration*int(time.Second)).String())
+	m.populateHeader(m.album.Name, subtitle, m.album.Duration, m.album.CoverArt)
+	playBtn := m.drawPlaylistAlbumButtons(m.album.Song)
+	m.populateSongs(m.album.Song, playBtn)
+
+}
+
+func (m *main) drawArtist() {
+	m.populateHeader(m.artist.Name, m.artistInfo.Biography, 0, m.artist.CoverArt)
+	btn := m.drawArtistButtons()
+	m.populateAlbums(btn)
+}
+
+func (m *main) drawSearch() {
+	sub := fmt.Sprintf("Query: %s", m.query)
+	m.populateHeader("Search Results", sub, 0, "")
+	m.populateSearchResults()
+}
+
+func (m *main) Update() {
+	m.view.Clear()
+	switch m.mode {
+	case mainModeAlbum:
+		m.drawAlbum()
+	case mainModePlaylist:
+		m.drawPlaylist()
+	case mainModeArtist:
+		m.drawArtist()
+	case mainModeSearch:
+		m.drawSearch()
+	}
+	m.songList.Focus(nil)
+}
+
+func (m *main) drawArtistButtons() *tview.Button {
+	// Add buttons: Radio
+	songs, _ := m.client.GetTopSongs(m.artist.Name, 10)
+	f := tview.NewFlex()
+	f.SetBackgroundColor(tcell.ColorBlack)
+	f.SetBorderPadding(0, 0, 2, 2)
+	// Buttons
+	radio := tview.NewButton("Radio")
+	top10 := tview.NewButton("Top 10")
+	// Button callbacks
+	top10.SetSelectedFunc(func() {
+		top10.Blur()
+		m.log("Playing %s's top 10 songs", m.artist.Name)
+		m.playAllFunc(songs...)
+		m.songList.Focus(nil)
+	})
+	radio.SetSelectedFunc(func() {
+		radio.Blur()
+		m.log("Generating %s's radio", m.artist.Name)
+		go func() {
+			songs := []*subsonic.Child{}
+			if config.ExperimentalRadioAlgo() {
+				songs, _ = m.client.GetExperimentalArtistRadio(m.artist, m.artistInfo, config.MaxRadioSongs())
+			}
+			if len(songs) == 0 {
+				songs, _ = m.client.GetSimilarSongs(m.artist.ID, config.MaxRadioSongs())
+			}
+
+			common.ShuffleSlice(songs)
+			m.playAllFunc(songs...)
+		}()
+		m.songList.Focus(nil)
+	})
+	radio.SetInputCapture(func(event *tcell.EventKey) *tcell.EventKey {
+		switch event.Key() {
+		case tcell.KeyDown:
+			radio.Blur()
+			m.songList.Focus(nil)
+			return nil
+		case tcell.KeyRight:
+			radio.Blur()
+			top10.Focus(nil)
+			return nil
+		}
+		return event
+	})
+	top10.SetInputCapture(func(event *tcell.EventKey) *tcell.EventKey {
+		switch event.Key() {
+		case tcell.KeyDown:
+			top10.Blur()
+			m.songList.Focus(nil)
+			return nil
+		case tcell.KeyLeft:
+			top10.Blur()
+			radio.Focus(nil)
+			return nil
+		}
+		return event
+	})
+
+	f.AddItem(radio, 0, 1, false)
+	f.AddItem(tview.NewBox(), 0, 1, false)
+	if len(songs) > 0 {
+		f.AddItem(top10, 0, 1, false)
+	}
+	f.AddItem(tview.NewBox(), 0, 1, false)
+
+	// Add the buttons to the view
+	m.view.AddItem(f, 1, 0, false)
+	// Margin bottom of 1 line
+	m.view.AddItem(tview.NewBox(), 1, 0, false)
+
+	return radio
+}
+
+func (m *main) populateAlbums(btn *tview.Button) {
+	table := tview.NewTable()
+	table.SetWrapSelection(true, false)
+	for i, album := range m.artist.Album {
+		year := tview.NewTableCell(fmt.Sprintf("%d", album.Year)).SetTextColor(tcell.ColorYellow)
+		name := tview.NewTableCell(album.Name).SetExpansion(2).SetAlign(tview.AlignCenter)
+		d := time.Second * time.Duration(album.Duration)
+		duration := tview.NewTableCell(d.String()).SetAlign(tview.AlignRight).SetExpansion(1)
+
+		table.SetCell(i, 0, year)
+		table.SetCell(i, 1, name)
+		table.SetCell(i, 2, duration)
+	}
+
+	table.SetFocusFunc(func() {
+		m.view.SetBorderColor(tcell.ColorRed)
+		table.SetSelectable(true, false)
+	})
+	table.SetBlurFunc(func() {
+		m.view.SetBorderColor(tcell.ColorWhite)
+		table.SetSelectable(false, false)
+	})
+	table.SetSelectedFunc(func(row, column int) {
+		alb, _ := m.client.GetAlbum(m.artist.Album[row].ID)
+		m.SetAlbum(alb)
+		m.view.Focus(nil)
+	})
+	table.SetInputCapture(func(event *tcell.EventKey) *tcell.EventKey {
+		row, _ := m.songList.GetSelection()
+		if row == 0 && event.Key() == tcell.KeyUp {
+			table.Blur()
+			m.view.SetBorderColor(tcell.ColorRed)
+			btn.Focus(nil)
+			return nil
+		}
+		return event
+	})
+	m.songList = table
+
+	m.view.AddItem(table, 0, 1, false)
+
+}
+
+func (m *main) drawPlaylistAlbumButtons(songs []*subsonic.Child) *tview.Button {
+	// Add buttons: Play | Shuffle | Add to queue
+	f := tview.NewFlex()
+	f.SetBackgroundColor(tcell.ColorBlack)
+	f.SetBorderPadding(0, 0, 2, 2)
+	// Buttons
+	play := tview.NewButton("Play")
+	shuffle := tview.NewButton("Shuffle")
+	queue := tview.NewButton("Queue")
+	artist := tview.NewButton("Artist")
+	// Button callbacks
+	play.SetSelectedFunc(func() {
+		play.Blur()
+		m.playAllFunc(songs...)
+		m.songList.Focus(nil)
+	})
+	play.SetInputCapture(func(event *tcell.EventKey) *tcell.EventKey {
+		switch event.Key() {
+		case tcell.KeyLeft:
+			play.Blur()
+			artist.Focus(nil)
+			return nil
+		case tcell.KeyRight:
+			play.Blur()
+			shuffle.Focus(nil)
+			return nil
+		case tcell.KeyDown:
+			play.Blur()
+			m.songList.Focus(nil)
+			return nil
+		}
+		return event
+	})
+	shuffle.SetSelectedFunc(func() {
+		shuffle.Blur()
+		cpy := make([]*subsonic.Child, len(songs))
+		copy(cpy, songs)
+		common.ShuffleSlice(cpy)
+		m.playAllFunc(cpy...)
+		m.songList.Focus(nil)
+	})
+	shuffle.SetInputCapture(func(event *tcell.EventKey) *tcell.EventKey {
+		switch event.Key() {
+		case tcell.KeyLeft:
+			shuffle.Blur()
+			play.Focus(nil)
+			return nil
+		case tcell.KeyRight:
+			shuffle.Blur()
+			queue.Focus(nil)
+			return nil
+		case tcell.KeyDown:
+			shuffle.Blur()
+			m.songList.Focus(nil)
+			return nil
+		}
+		return event
+	})
+	queue.SetSelectedFunc(func() {
+		queue.Blur()
+		m.addSongsFunc(songs...)
+		m.songList.Focus(nil)
+	})
+	queue.SetInputCapture(func(event *tcell.EventKey) *tcell.EventKey {
+		switch event.Key() {
+		case tcell.KeyLeft:
+			queue.Blur()
+			shuffle.Focus(nil)
+			return nil
+		case tcell.KeyRight:
+			queue.Blur()
+			artist.Focus(nil)
+			return nil
+		case tcell.KeyDown:
+			queue.Blur()
+			m.songList.Focus(nil)
+			return nil
+		}
+		return event
+	})
+	artist.SetSelectedFunc(func() {
+		artist.Blur()
+		ar, _ := m.client.GetArtist(m.album.ArtistID)
+		m.SetArtist(ar)
+		m.songList.Focus(nil)
+	})
+	artist.SetInputCapture(func(event *tcell.EventKey) *tcell.EventKey {
+		switch event.Key() {
+		case tcell.KeyLeft:
+			artist.Blur()
+			queue.Focus(nil)
+			return nil
+		case tcell.KeyRight:
+			artist.Blur()
+			play.Focus(nil)
+			return nil
+		case tcell.KeyDown:
+			artist.Blur()
+			m.songList.Focus(nil)
+			return nil
+		}
+		return event
+	})
+
+	f.AddItem(play, 0, 1, true)
+	f.AddItem(tview.NewBox(), 0, 1, false)
+	f.AddItem(shuffle, 0, 1, false)
+	f.AddItem(tview.NewBox(), 0, 1, false)
+	f.AddItem(queue, 0, 1, false)
+	f.AddItem(tview.NewBox(), 0, 1, false)
+	f.AddItem(artist, 0, 1, false)
+
+	// Add the buttons to the view
+	m.view.AddItem(f, 1, 0, false)
+	// Margin bottom of 1 line
+	m.view.AddItem(tview.NewBox(), 1, 0, false)
+
+	return play
+}
+
+func (m *main) populateSongs(songs []*subsonic.Child, play *tview.Button) {
+	table := tview.NewTable()
+	table.SetWrapSelection(true, false)
+	for i, song := range songs {
+		num := tview.NewTableCell(fmt.Sprintf("%d", i+1)).SetTextColor(tcell.ColorYellow)
+		title := tview.NewTableCell(song.Title).SetMaxWidth(15).SetExpansion(2)
+		artist := tview.NewTableCell(song.Artist).SetMaxWidth(15).SetExpansion(1)
+		d := time.Second * time.Duration(song.Duration)
+		duration := tview.NewTableCell(d.String()).SetAlign(tview.AlignRight).SetExpansion(1)
+		table.SetCell(i, 0, num)
+		table.SetCell(i, 1, title)
+		table.SetCell(i, 2, artist)
+		table.SetCell(i, 3, duration)
+	}
+
+	m.songList = table
+	table.SetInputCapture(func(event *tcell.EventKey) *tcell.EventKey {
+		// on first line and pressing up -> play button
+		row, _ := table.GetSelection()
+		if row == 0 && event.Key() == tcell.KeyUp {
+			table.Blur()
+			m.view.SetBorderColor(tcell.ColorRed)
+			play.Focus(nil)
+			return nil
+		} else if event.Rune() == 'r' {
+			song := songs[row]
+			m.log("Generating song (%s) radio....", song.Title)
+			go func() {
+				radioSongs, _ := m.client.GetSimilarSongs(song.ID, config.MaxRadioSongs())
+				m.playAllFunc(radioSongs...)
+			}()
+		}
+		return event
+	})
+	table.SetFocusFunc(func() {
+		m.view.SetBorderColor(tcell.ColorRed)
+		table.SetSelectable(true, false)
+	})
+	table.SetBlurFunc(func() {
+		m.view.SetBorderColor(tcell.ColorWhite)
+		table.SetSelectable(false, false)
+	})
+	m.songList.SetSelectedFunc(func(row, column int) {
+		m.addSongsFunc(songs[row])
+	})
+
+	m.view.AddItem(table, 0, 1, false)
+}
+
+func (m *main) populateSearchResults() {
+	table := tview.NewTable()
+	table.SetWrapSelection(true, false)
+	row := 0
+	lastArtist := 0
+	lastAlbum := 0
+
+	// Artists
+	if len(m.searchResult.Artist) > 0 {
+		// Header
+		header := tview.NewTableCell("Artists").SetSelectable(false)
+		table.SetCell(row, 0, header)
+		row++
+		//List
+		for i, artist := range m.searchResult.Artist {
+			index := tview.NewTableCell(fmt.Sprintf("%d", i+1)).
+				SetTextColor(tcell.ColorYellow)
+			a := tview.NewTableCell(artist.Name).SetExpansion(2).SetMaxWidth(15)
+			acount := tview.NewTableCell(fmt.Sprintf("%d Albums", artist.AlbumCount)).
+				SetExpansion(1).SetAlign(tview.AlignRight).SetMaxWidth(15)
+			table.SetCell(row, 0, index)
+			table.SetCell(row, 1, a)
+			table.SetCell(row, 2, acount)
+			row++
+		}
+		lastArtist = row
+	}
+
+	// Albums
+
+	if len(m.searchResult.Album) > 0 {
+		// Header
+		header := tview.NewTableCell("Albums").SetSelectable(false)
+		table.SetCell(row, 0, header)
+		row++
+		//List
+		for i, album := range m.searchResult.Album {
+			index := tview.NewTableCell(fmt.Sprintf("%d", i+1)).
+				SetTextColor(tcell.ColorYellow)
+			title := tview.NewTableCell(album.Name).SetExpansion(2).SetMaxWidth(15)
+			artist := tview.NewTableCell(album.Artist).
+				SetExpansion(1).SetAlign(tview.AlignRight).SetMaxWidth(15)
+			table.SetCell(row, 0, index)
+			table.SetCell(row, 1, title)
+			table.SetCell(row, 2, artist)
+			row++
+		}
+		lastAlbum = row
+	}
+	// Songs
+	if len(m.searchResult.Song) > 0 {
+		// Header
+		header := tview.NewTableCell("Songs").SetSelectable(false)
+		table.SetCell(row, 0, header)
+		row++
+		//List
+		for i, song := range m.searchResult.Song {
+			index := tview.NewTableCell(fmt.Sprintf("%d", i+1)).
+				SetTextColor(tcell.ColorYellow)
+			title := tview.NewTableCell(song.Title).SetExpansion(2).SetMaxWidth(15)
+			artist := tview.NewTableCell(song.Artist).
+				SetExpansion(1).SetAlign(tview.AlignRight).SetMaxWidth(15)
+			table.SetCell(row, 0, index)
+			table.SetCell(row, 1, title)
+			table.SetCell(row, 2, artist)
+			row++
+		}
+	}
+
+	m.songList = table
+	table.SetFocusFunc(func() {
+		m.view.SetBorderColor(tcell.ColorRed)
+		table.SetSelectable(true, false)
+	})
+	table.SetBlurFunc(func() {
+		m.view.SetBorderColor(tcell.ColorWhite)
+		table.SetSelectable(false, false)
+	})
+	table.SetInputCapture(func(event *tcell.EventKey) *tcell.EventKey {
+		row, _ := table.GetSelection()
+		if row <= lastAlbum {
+			return event
+		}
+		if event.Rune() != 'r' {
+			return event
+		}
+		cell := table.GetCell(row, 0)
+		index, err := strconv.Atoi(cell.Text)
+		if err != nil {
+			return nil
+		}
+		song := m.searchResult.Song[index-1]
+		m.log("Generating song (%s) radio....", song.Title)
+		go func() {
+			radioSongs, _ := m.client.GetSimilarSongs(song.ID, config.MaxRadioSongs())
+			m.playAllFunc(radioSongs...)
+		}()
+		return nil
+	})
+	m.songList.SetSelectedFunc(func(row, column int) {
+		cell := table.GetCell(row, 0)
+		index, err := strconv.Atoi(cell.Text)
+		if err != nil {
+			return
+		}
+		if row <= lastArtist {
+			artist, _ := m.client.GetArtist(m.searchResult.Artist[index-1].ID)
+			m.SetArtist(artist)
+		} else if lastArtist < row && row <= lastAlbum {
+			album, _ := m.client.GetAlbum(m.searchResult.Album[index-1].ID)
+			m.SetAlbum(album)
+		} else {
+			m.addSongsFunc(m.searchResult.Song[index-1])
+		}
+	})
+
+	m.view.AddItem(table, 0, 1, false)
+}
+
+func (m *main) populateHeader(title, subtitle string,
+	duration int, coverArtID string) {
+
+	header := tview.NewFlex()
+	art := tview.NewImage()
+
+	img, _ := m.client.GetCoverArt(coverArtID)
+	art.SetImage(img)
+	t := tview.NewTextView().
+		SetTextColor(tcell.ColorLightCyan).
+		SetTextAlign(tview.AlignCenter).
+		SetText(title)
+	s := tview.NewTextView().
+		SetTextColor(tcell.ColorYellow).
+		SetTextAlign(tview.AlignCenter).
+		SetText(subtitle).SetWordWrap(true)
+
+	g := tview.NewFlex().SetDirection(tview.FlexRow)
+	g.AddItem(tview.NewBox(), 1, 1, false)
+	g.AddItem(t, 1, 1, false)
+	g.AddItem(s, 0, 1, false)
+
+	header.AddItem(art, 0, 1, false)
+	header.AddItem(g, 0, 3, false)
+	size := 6
+	if m.miniView {
+		size = 4
+	}
+	m.view.AddItem(header, size, 1, false)
+
+	// Margin bottom of 1 line
+	if !m.miniView {
+		m.view.AddItem(tview.NewBox(), 1, 0, false)
+	}
+}
+
+func (m *main) SetPlayAllFunc(f func(song ...*subsonic.Child)) {
+	m.playAllFunc = f
+}
+
+func (m *main) SetPlayAddSongFunc(f func(song ...*subsonic.Child)) {
+	m.addSongsFunc = f
+}
+
+func (m *main) GetView() tview.Primitive {
+	return m.view
+}

internal/tui/views/player.go

@@ -0,0 +1,113 @@
+package views
+
+import (
+	"fmt"
+	"time"
+
+	"git.dayanhub.com/sagi/subsonic-tui/internal/client"
+	"github.com/delucks/go-subsonic"
+	"github.com/gdamore/tcell/v2"
+	"github.com/rivo/tview"
+)
+
+var _ View = &player{}
+
+type player struct {
+	client   *client.Client
+	view     tview.Primitive
+	grid     *tview.Grid
+	artwork  *tview.Image
+	progress *tview.Flex
+	songInfo *tview.TextView
+	song     *subsonic.Child
+}
+
+func NewPlayer(client *client.Client) *player {
+	grid := tview.NewGrid().SetColumns(9, 0)
+
+	//album art
+	art := tview.NewImage()
+
+	grid.AddItem(art, 0, 0, 1, 1, 4, 4, false)
+	// Progress
+	progress := tview.NewFlex()
+
+	// Song songInfo
+	songInfo := tview.NewTextView()
+	songInfo.SetDynamicColors(true)
+
+	// Info + Progress
+	songProg := tview.NewFlex().SetDirection(tview.FlexRow)
+	songProg.AddItem(songInfo, 0, 4, false)
+	songProg.AddItem(progress, 0, 1, false)
+	grid.AddItem(songProg, 0, 1, 1, 1, 0, 0, false)
+
+	player := &player{
+		client:   client,
+		view:     grid,
+		artwork:  art,
+		songInfo: songInfo,
+		grid:     grid,
+		progress: progress,
+	}
+
+	player.SetSongInfo(&subsonic.Child{
+		Title:    "Subsonic TUI",
+		Album:    "MaVeZe",
+		Artist:   "ZeGoomba",
+		CoverArt: "",
+		Duration: 0,
+	})
+	player.UpdateProgress(time.Duration(0))
+
+	return player
+}
+
+func (p *player) SetSongInfo(song *subsonic.Child) {
+	info := fmt.Sprintf("[yellow]Title:[white] %s\n[yellow]Album:[white] %s\n[yellow]Atrists:[white] %s",
+		song.Title, song.Album, song.Artist)
+	p.songInfo.SetText(info)
+	p.song = song
+	p.LoadAlbumArt(song.CoverArt)
+}
+
+func (p *player) LoadAlbumArt(ID string) {
+	i, _ := p.client.GetCoverArt(ID)
+	p.artwork.SetImage(i)
+}
+
+func (p *player) UpdateProgress(elapsed time.Duration) {
+
+	if p.song.Duration == 0 {
+		// Startup... Show version number
+		versionInfo := tview.NewTextView().SetText("Version: 0.1")
+		versionInfo.SetBackgroundColor(tcell.ColorDarkGray)
+		versionInfo.SetTextColor(tcell.ColorDarkRed)
+		p.progress.AddItem(versionInfo, 0, 1, false)
+		return
+	}
+
+	songDuration := time.Duration(p.song.Duration) * time.Second
+	overlappedBox := tview.NewTextView()
+	overlappedBox.SetBackgroundColor(tcell.ColorDarkRed)
+	overlappedBox.SetTextColor(tcell.ColorDarkGray)
+	overlappedBox.SetText(songDuration.String())
+	remainingBox := tview.NewTextView()
+	remainingBox.SetBackgroundColor(tcell.ColorDarkGray)
+	remainingBox.SetTextColor(tcell.ColorDarkRed)
+	remainingBox.SetTextAlign(tview.AlignRight)
+	rm := time.Duration(songDuration.Seconds()-elapsed.Seconds()) * time.Second
+	remaining := fmt.Sprintf("-%s", rm.String())
+	remainingBox.SetText(remaining)
+	p.progress.Clear()
+	p.progress.AddItem(overlappedBox, 0, int(elapsed.Seconds()), false)
+	p.progress.AddItem(remainingBox, 0, int(songDuration.Seconds())-int(elapsed.Seconds()), false)
+}
+
+func (p *player) GetView() tview.Primitive {
+	return p.view
+}
+
+func (p *player) Update() {
+	//p.UpdateProgress("00:00", 50)
+}

internal/tui/views/playlists.go

@@ -0,0 +1,63 @@
+package views
+
+import (
+	"git.dayanhub.com/sagi/subsonic-tui/internal/client"
+	"github.com/delucks/go-subsonic"
+	"github.com/gdamore/tcell/v2"
+	"github.com/rivo/tview"
+)
+
+var _ View = &playlists{}
+
+type playlists struct {
+	view     *tview.Table
+	client   *client.Client
+	callback func(playlist *subsonic.Playlist)
+}
+
+func NewPlaylists(client *client.Client) *playlists {
+
+	obj := &playlists{
+		client: client,
+	}
+
+	list := tview.NewTable()
+
+	list.SetBackgroundColor(tcell.ColorBlack)
+	list.SetTitle("Playlists [3]")
+	list.SetBorder(true)
+	list.SetFocusFunc(func() {
+		list.SetBorderColor(tcell.ColorRed)
+		list.SetSelectable(true, false)
+	})
+	list.SetBlurFunc(func() {
+		list.SetBorderColor(tcell.ColorWhite)
+		list.SetSelectable(false, false)
+	})
+
+	pls, _ := client.GetPlaylists()
+	for i, pl := range pls {
+		cell := tview.NewTableCell(pl.Name).SetExpansion(1)
+		list.SetCell(i, 0, cell)
+	}
+
+	list.SetSelectedFunc(func(row, column int) {
+		obj.callback(pls[row])
+	})
+
+	obj.view = list
+
+	return obj
+}
+
+func (p *playlists) SetCallback(f func(playlist *subsonic.Playlist)) {
+	p.callback = f
+}
+
+func (p *playlists) Update() {
+
+}
+
+func (p *playlists) GetView() tview.Primitive {
+	return p.view
+}

internal/tui/views/queue.go

@@ -0,0 +1,131 @@
+package views
+
+import (
+	"fmt"
+	"time"
+
+	"github.com/delucks/go-subsonic"
+	"github.com/gdamore/tcell/v2"
+	"github.com/rivo/tview"
+)
+
+var _ View = &queue{}
+
+type queue struct {
+	view        *tview.Table
+	currentSong int
+	songQueue   []*subsonic.Child
+	playFunc    func(song *subsonic.Child)
+}
+
+func NewQueue() *queue {
+
+	table := tview.NewTable()
+
+	table.SetBackgroundColor(tcell.ColorBlack)
+	table.SetTitle("Queue [4]")
+	table.SetBorder(true)
+	table.SetFocusFunc(func() {
+		table.SetBorderColor(tcell.ColorRed)
+	})
+	table.SetBlurFunc(func() {
+		table.SetBorderColor(tcell.ColorWhite)
+	})
+
+	return &queue{
+		view:        table,
+		currentSong: 0,
+	}
+
+}
+
+func (q *queue) SetPlayFunc(f func(song *subsonic.Child)) {
+	q.playFunc = f
+}
+
+func (q *queue) PlayCurrent() {
+	if len(q.songQueue) > 0 {
+		q.playFunc(q.songQueue[q.currentSong])
+	}
+}
+
+func (q *queue) Add(songs ...*subsonic.Child) {
+	shouldStartPlaying := len(q.songQueue) == 0
+	q.songQueue = append(q.songQueue, songs...)
+	if shouldStartPlaying {
+		q.currentSong = 0
+		q.Update()
+		q.playFunc(songs[0])
+	}
+}
+
+func (q *queue) ReplaceAll(songs ...*subsonic.Child) {
+	q.Clear()
+	q.Add(songs...)
+	q.view.ScrollToBeginning()
+
+}
+
+func (q *queue) Next() {
+	if len(q.songQueue) > q.currentSong+1 {
+		q.currentSong = q.currentSong + 1
+		song := q.songQueue[q.currentSong]
+		q.playFunc(song)
+	}
+
+}
+func (q *queue) Prev() {
+	if q.currentSong > 0 {
+		q.currentSong = q.currentSong - 1
+		song := q.songQueue[q.currentSong]
+		q.Update()
+		q.playFunc(song)
+	}
+
+}
+
+func (q *queue) Clear() {
+	q.songQueue = []*subsonic.Child{}
+	q.Update()
+}
+
+func (q *queue) drawQueue() {
+	q.view.Clear()
+	list := q.view
+	list.SetWrapSelection(true, false)
+	list.SetSelectable(true, false)
+	for i, song := range q.songQueue {
+		isCurrentSong := q.currentSong == i
+		isPlayed := i < q.currentSong
+		bgColor := tcell.ColorBlack
+		if isCurrentSong {
+			bgColor = tcell.ColorDarkRed
+		} else if isPlayed {
+			bgColor = tcell.ColorDarkGray
+		}
+		num := tview.NewTableCell(fmt.Sprintf("%d", i+1)).SetTextColor(tcell.ColorYellow).SetBackgroundColor(bgColor)
+		title := tview.NewTableCell(song.Title).SetMaxWidth(15).SetExpansion(2).SetBackgroundColor(bgColor)
+		artist := tview.NewTableCell(song.Artist).SetMaxWidth(15).SetExpansion(1).SetBackgroundColor(bgColor)
+		d := time.Second * time.Duration(song.Duration)
+		duration := tview.NewTableCell(d.String()).SetAlign(tview.AlignRight).SetExpansion(1).SetBackgroundColor(bgColor)
+		list.SetCell(i, 0, num)
+		list.SetCell(i, 1, title)
+		list.SetCell(i, 2, artist)
+		list.SetCell(i, 3, duration)
+	}
+
+	list.SetSelectedFunc(func(row, column int) {
+		q.currentSong = row
+		q.playFunc(q.songQueue[row])
+	})
+}
+
+func (q *queue) Update() {
+
+	q.drawQueue()
+
+}
+
+func (q *queue) GetView() tview.Primitive {
+	return q.view
+}

internal/tui/views/status_line.go

@@ -0,0 +1,93 @@
+package views
+
+import (
+	"fmt"
+
+	"github.com/gdamore/tcell/v2"
+	"github.com/rivo/tview"
+)
+
+var _ View = &statusLine{}
+
+type statusLine struct {
+	view         *tview.Flex
+	mode         Statusmode
+	onUpdateFunc func()
+	onSearchFunc func(quary string)
+}
+
+type Statusmode int
+
+const (
+	StatusModeLog    Statusmode = iota
+	StatusModeSearch Statusmode = iota
+)
+
+func NewStatusLine() *statusLine {
+	status := tview.NewFlex()
+	// Default empty box
+	status.AddItem(tview.NewBox(), 0, 1, false)
+
+	return &statusLine{
+		view: status,
+		mode: StatusModeLog,
+	}
+}
+
+func (s *statusLine) SetSearchFunc(f func(quary string)) {
+	s.onSearchFunc = f
+}
+
+func (s *statusLine) Mode() Statusmode {
+	return s.mode
+}
+
+func (s *statusLine) Search() {
+	s.mode = StatusModeSearch
+	s.view.Clear()
+	label := "Search: "
+	_, _, w, _ := s.view.GetRect()
+	query := ""
+	inputField := tview.NewInputField()
+	inputField.
+		SetLabel(label).
+		SetFieldWidth(w - len(label)).
+		SetDoneFunc(func(key tcell.Key) {
+			if key == tcell.KeyEnter {
+				s.mode = StatusModeLog
+				s.onSearchFunc(query)
+			} else if key == tcell.KeyEsc {
+				s.mode = StatusModeLog
+				s.onSearchFunc("")
+			}
+		}).
+		SetChangedFunc(func(text string) {
+			query = text
+		})
+	inputField.Focus(nil)
+	s.view.AddItem(inputField, 0, 1, true)
+	s.Update()
+}
+
+func (s *statusLine) SetOnUpdateFunc(f func()) {
+	s.onUpdateFunc = f
+}
+
+func (s *statusLine) Log(format string, a ...any) {
+	if s.mode != StatusModeLog {
+		return
+	}
+	str := fmt.Sprintf(format, a...)
+	s.view.Clear()
+	txt := tview.NewTextView().SetDynamicColors(true)
+	txt.SetText(str)
+	s.view.AddItem(txt, 0, 1, false)
+	s.Update()
+}
+
+func (s *statusLine) GetView() tview.Primitive {
+	return s.view
+}
+func (s *statusLine) Update() {
+	s.onUpdateFunc()
+}

internal/tui/views/view.go

@@ -0,0 +1,10 @@
+package views
+
+import "github.com/rivo/tview"
+
+type ViewEvent string
+
+type View interface {
+	GetView() tview.Primitive
+	Update()
+}

main.go

@@ -0,0 +1,40 @@
+package main
+
+import (
+	"fmt"
+
+	"git.dayanhub.com/sagi/subsonic-tui/internal/client"
+	"git.dayanhub.com/sagi/subsonic-tui/internal/config"
+	"git.dayanhub.com/sagi/subsonic-tui/internal/playback"
+	"git.dayanhub.com/sagi/subsonic-tui/internal/tui"
+)
+
+func main() {
+	// Create Client
+	c := client.NewClient(config.URL())
+	err := c.Authenticate(config.Username(), config.Password())
+	if err != nil {
+		// We need to show Login...
+		login := tui.NewLogin()
+		err := login.Run()
+		if err != nil {
+			panic(err)
+		}
+
+	}
+	fmt.Println("Trying to login...")
+	c = client.NewClient(config.URL())
+	err = c.Authenticate(config.Username(), config.Password())
+	if err != nil {
+		panic(err)
+	}
+	// Saving config - will result in adding new defaults to the file
+	config.SaveConfig()
+	playbackCtl := playback.NewController()
+	defer playbackCtl.Close()
+
+	tui := tui.NewPlayer(c, playbackCtl)
+	if err := tui.Run(); err != nil {
+		panic(err)
+	}
+}

vendor/github.com/creasty/defaults/.gitignore

@@ -0,0 +1 @@
+.DS_Store

vendor/github.com/creasty/defaults/LICENSE

@@ -0,0 +1,22 @@
+Copyright (c) 2017-present Yuki Iwanaga
+
+MIT License
+
+Permission is hereby granted, free of charge, to any person obtaining
+a copy of this software and associated documentation files (the
+"Software"), to deal in the Software without restriction, including
+without limitation the rights to use, copy, modify, merge, publish,
+distribute, sublicense, and/or sell copies of the Software, and to
+permit persons to whom the Software is furnished to do so, subject to
+the following conditions:
+
+The above copyright notice and this permission notice shall be
+included in all copies or substantial portions of the Software.
+
+THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
+EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
+MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND
+NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE
+LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION
+OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
+WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.

vendor/github.com/creasty/defaults/Makefile

@@ -0,0 +1,29 @@
+SHELL := /bin/bash -eu -o pipefail
+
+GO_TEST_FLAGS  := -v
+
+PACKAGE_DIRS := $(shell go list ./... 2> /dev/null | grep -v /vendor/)
+SRC_FILES    := $(shell find . -name '*.go' -not -path './vendor/*')
+
+
+#  Tasks
+#-----------------------------------------------
+.PHONY: lint
+lint:
+	@gofmt -e -d -s $(SRC_FILES) | awk '{ e = 1; print $0 } END { if (e) exit(1) }'
+	@golangci-lint --disable errcheck,unused run
+
+.PHONY: test
+test: lint
+	@go test $(GO_TEST_FLAGS) $(PACKAGE_DIRS)
+
+.PHONY: ci-test
+ci-test: lint
+	@echo > coverage.txt
+	@for d in $(PACKAGE_DIRS); do \
+		go test -coverprofile=profile.out -covermode=atomic -race -v $$d; \
+		if [ -f profile.out ]; then \
+			cat profile.out >> coverage.txt; \
+			rm profile.out; \
+		fi; \
+	done

vendor/github.com/creasty/defaults/README.md

@@ -0,0 +1,75 @@
+defaults
+========
+
+[![CircleCI](https://circleci.com/gh/creasty/defaults/tree/master.svg?style=svg)](https://circleci.com/gh/creasty/defaults/tree/master)
+[![codecov](https://codecov.io/gh/creasty/defaults/branch/master/graph/badge.svg)](https://codecov.io/gh/creasty/defaults)
+[![GitHub release](https://img.shields.io/github/release/creasty/defaults.svg)](https://github.com/creasty/defaults/releases)
+[![License](https://img.shields.io/github/license/creasty/defaults.svg)](./LICENSE)
+
+Initialize structs with default values
+
+- Supports almost all kind of types
+  - Scalar types
+    - `int/8/16/32/64`, `uint/8/16/32/64`, `float32/64`
+    - `uintptr`, `bool`, `string`
+  - Complex types
+    - `map`, `slice`, `struct`
+  - Nested types
+    - `map[K1]map[K2]Struct`, `[]map[K1]Struct[]`
+  - Aliased types
+    - `time.Duration`
+    - e.g., `type Enum string`
+  - Pointer types
+    - e.g., `*SampleStruct`, `*int`
+- Recursively initializes fields in a struct
+- Dynamically sets default values by [`defaults.Setter`](./setter.go) interface
+- Preserves non-initial values from being reset with a default value
+
+
+Usage
+-----
+
+```go
+type Gender string
+
+type Sample struct {
+	Name   string `default:"John Smith"`
+	Age    int    `default:"27"`
+	Gender Gender `default:"m"`
+
+	Slice       []string       `default:"[]"`
+	SliceByJSON []int          `default:"[1, 2, 3]"` // Supports JSON
+
+	Map                 map[string]int `default:"{}"`
+	MapByJSON           map[string]int `default:"{\"foo\": 123}"`
+	MapOfStruct         map[string]OtherStruct
+	MapOfPtrStruct      map[string]*OtherStruct
+	MapOfStructWithTag  map[string]OtherStruct `default:"{\"Key1\": {\"Foo\":123}}"`
+    
+	Struct    OtherStruct  `default:"{}"`
+	StructPtr *OtherStruct `default:"{\"Foo\": 123}"`
+
+	NoTag  OtherStruct               // Recurses into a nested struct by default
+	OptOut OtherStruct `default:"-"` // Opt-out
+}
+
+type OtherStruct struct {
+	Hello  string `default:"world"` // Tags in a nested struct also work
+	Foo    int    `default:"-"`
+	Random int    `default:"-"`
+}
+
+// SetDefaults implements defaults.Setter interface
+func (s *OtherStruct) SetDefaults() {
+	if defaults.CanUpdate(s.Random) { // Check if it's a zero value (recommended)
+		s.Random = rand.Int() // Set a dynamic value
+	}
+}
+```
+
+```go
+obj := &Sample{}
+if err := defaults.Set(obj); err != nil {
+	panic(err)
+}
+```

vendor/github.com/creasty/defaults/defaults.go

@@ -0,0 +1,244 @@
+package defaults
+
+import (
+	"encoding"
+	"encoding/json"
+	"errors"
+	"reflect"
+	"strconv"
+	"time"
+)
+
+var (
+	errInvalidType = errors.New("not a struct pointer")
+)
+
+const (
+	fieldName = "default"
+)
+
+// Set initializes members in a struct referenced by a pointer.
+// Maps and slices are initialized by `make` and other primitive types are set with default values.
+// `ptr` should be a struct pointer
+func Set(ptr interface{}) error {
+	if reflect.TypeOf(ptr).Kind() != reflect.Ptr {
+		return errInvalidType
+	}
+
+	v := reflect.ValueOf(ptr).Elem()
+	t := v.Type()
+
+	if t.Kind() != reflect.Struct {
+		return errInvalidType
+	}
+
+	for i := 0; i < t.NumField(); i++ {
+		if defaultVal := t.Field(i).Tag.Get(fieldName); defaultVal != "-" {
+			if err := setField(v.Field(i), defaultVal); err != nil {
+				return err
+			}
+		}
+	}
+	callSetter(ptr)
+	return nil
+}
+
+// MustSet function is a wrapper of Set function
+// It will call Set and panic if err not equals nil.
+func MustSet(ptr interface{}) {
+	if err := Set(ptr); err != nil {
+		panic(err)
+	}
+}
+
+func setField(field reflect.Value, defaultVal string) error {
+	if !field.CanSet() {
+		return nil
+	}
+
+	if !shouldInitializeField(field, defaultVal) {
+		return nil
+	}
+
+	isInitial := isInitialValue(field)
+	if isInitial {
+		if unmarshalByInterface(field, defaultVal) {
+			return nil
+		}
+
+		switch field.Kind() {
+		case reflect.Bool:
+			if val, err := strconv.ParseBool(defaultVal); err == nil {
+				field.Set(reflect.ValueOf(val).Convert(field.Type()))
+			}
+		case reflect.Int:
+			if val, err := strconv.ParseInt(defaultVal, 0, strconv.IntSize); err == nil {
+				field.Set(reflect.ValueOf(int(val)).Convert(field.Type()))
+			}
+		case reflect.Int8:
+			if val, err := strconv.ParseInt(defaultVal, 0, 8); err == nil {
+				field.Set(reflect.ValueOf(int8(val)).Convert(field.Type()))
+			}
+		case reflect.Int16:
+			if val, err := strconv.ParseInt(defaultVal, 0, 16); err == nil {
+				field.Set(reflect.ValueOf(int16(val)).Convert(field.Type()))
+			}
+		case reflect.Int32:
+			if val, err := strconv.ParseInt(defaultVal, 0, 32); err == nil {
+				field.Set(reflect.ValueOf(int32(val)).Convert(field.Type()))
+			}
+		case reflect.Int64:
+			if val, err := time.ParseDuration(defaultVal); err == nil {
+				field.Set(reflect.ValueOf(val).Convert(field.Type()))
+			} else if val, err := strconv.ParseInt(defaultVal, 0, 64); err == nil {
+				field.Set(reflect.ValueOf(val).Convert(field.Type()))
+			}
+		case reflect.Uint:
+			if val, err := strconv.ParseUint(defaultVal, 0, strconv.IntSize); err == nil {
+				field.Set(reflect.ValueOf(uint(val)).Convert(field.Type()))
+			}
+		case reflect.Uint8:
+			if val, err := strconv.ParseUint(defaultVal, 0, 8); err == nil {
+				field.Set(reflect.ValueOf(uint8(val)).Convert(field.Type()))
+			}
+		case reflect.Uint16:
+			if val, err := strconv.ParseUint(defaultVal, 0, 16); err == nil {
+				field.Set(reflect.ValueOf(uint16(val)).Convert(field.Type()))
+			}
+		case reflect.Uint32:
+			if val, err := strconv.ParseUint(defaultVal, 0, 32); err == nil {
+				field.Set(reflect.ValueOf(uint32(val)).Convert(field.Type()))
+			}
+		case reflect.Uint64:
+			if val, err := strconv.ParseUint(defaultVal, 0, 64); err == nil {
+				field.Set(reflect.ValueOf(val).Convert(field.Type()))
+			}
+		case reflect.Uintptr:
+			if val, err := strconv.ParseUint(defaultVal, 0, strconv.IntSize); err == nil {
+				field.Set(reflect.ValueOf(uintptr(val)).Convert(field.Type()))
+			}
+		case reflect.Float32:
+			if val, err := strconv.ParseFloat(defaultVal, 32); err == nil {
+				field.Set(reflect.ValueOf(float32(val)).Convert(field.Type()))
+			}
+		case reflect.Float64:
+			if val, err := strconv.ParseFloat(defaultVal, 64); err == nil {
+				field.Set(reflect.ValueOf(val).Convert(field.Type()))
+			}
+		case reflect.String:
+			field.Set(reflect.ValueOf(defaultVal).Convert(field.Type()))
+
+		case reflect.Slice:
+			ref := reflect.New(field.Type())
+			ref.Elem().Set(reflect.MakeSlice(field.Type(), 0, 0))
+			if defaultVal != "" && defaultVal != "[]" {
+				if err := json.Unmarshal([]byte(defaultVal), ref.Interface()); err != nil {
+					return err
+				}
+			}
+			field.Set(ref.Elem().Convert(field.Type()))
+		case reflect.Map:
+			ref := reflect.New(field.Type())
+			ref.Elem().Set(reflect.MakeMap(field.Type()))
+			if defaultVal != "" && defaultVal != "{}" {
+				if err := json.Unmarshal([]byte(defaultVal), ref.Interface()); err != nil {
+					return err
+				}
+			}
+			field.Set(ref.Elem().Convert(field.Type()))
+		case reflect.Struct:
+			if defaultVal != "" && defaultVal != "{}" {
+				if err := json.Unmarshal([]byte(defaultVal), field.Addr().Interface()); err != nil {
+					return err
+				}
+			}
+		case reflect.Ptr:
+			field.Set(reflect.New(field.Type().Elem()))
+		}
+	}
+
+	switch field.Kind() {
+	case reflect.Ptr:
+		if isInitial || field.Elem().Kind() == reflect.Struct {
+			setField(field.Elem(), defaultVal)
+			callSetter(field.Interface())
+		}
+	case reflect.Struct:
+		if err := Set(field.Addr().Interface()); err != nil {
+			return err
+		}
+	case reflect.Slice:
+		for j := 0; j < field.Len(); j++ {
+			if err := setField(field.Index(j), defaultVal); err != nil {
+				return err
+			}
+		}
+	case reflect.Map:
+		for _, e := range field.MapKeys() {
+			var v = field.MapIndex(e)
+
+			switch v.Kind() {
+			case reflect.Ptr:
+				switch v.Elem().Kind() {
+				case reflect.Struct, reflect.Slice, reflect.Map:
+					if err := setField(v.Elem(), ""); err != nil {
+						return err
+					}
+				}
+			case reflect.Struct, reflect.Slice, reflect.Map:
+				ref := reflect.New(v.Type())
+				ref.Elem().Set(v)
+				if err := setField(ref.Elem(), ""); err != nil {
+					return err
+				}
+				field.SetMapIndex(e, ref.Elem().Convert(v.Type()))
+			}
+		}
+	}
+
+	return nil
+}
+
+func unmarshalByInterface(field reflect.Value, defaultVal string) bool {
+	asText, ok := field.Addr().Interface().(encoding.TextUnmarshaler)
+	if ok && defaultVal != "" {
+		// if field implements encode.TextUnmarshaler, try to use it before decode by kind
+		if err := asText.UnmarshalText([]byte(defaultVal)); err == nil {
+			return true
+		}
+	}
+	asJSON, ok := field.Addr().Interface().(json.Unmarshaler)
+	if ok && defaultVal != "" && defaultVal != "{}" && defaultVal != "[]" {
+		// if field implements json.Unmarshaler, try to use it before decode by kind
+		if err := asJSON.UnmarshalJSON([]byte(defaultVal)); err == nil {
+			return true
+		}
+	}
+	return false
+}
+
+func isInitialValue(field reflect.Value) bool {
+	return reflect.DeepEqual(reflect.Zero(field.Type()).Interface(), field.Interface())
+}
+
+func shouldInitializeField(field reflect.Value, tag string) bool {
+	switch field.Kind() {
+	case reflect.Struct:
+		return true
+	case reflect.Ptr:
+		if !field.IsNil() && field.Elem().Kind() == reflect.Struct {
+			return true
+		}
+	case reflect.Slice:
+		return field.Len() > 0 || tag != ""
+	case reflect.Map:
+		return field.Len() > 0 || tag != ""
+	}
+
+	return tag != ""
+}
+
+// CanUpdate returns true when the given value is an initial value of its type
+func CanUpdate(v interface{}) bool {
+	return isInitialValue(reflect.ValueOf(v))
+}

vendor/github.com/creasty/defaults/setter.go

@@ -0,0 +1,12 @@
+package defaults
+
+// Setter is an interface for setting default values
+type Setter interface {
+	SetDefaults()
+}
+
+func callSetter(v interface{}) {
+	if ds, ok := v.(Setter); ok {
+		ds.SetDefaults()
+	}
+}

vendor/github.com/delucks/go-subsonic/.gitignore

@@ -0,0 +1,4 @@
+*.swp
+.idea
+subsonic-rest-api-1.16.1.xsd
+build/

vendor/github.com/delucks/go-subsonic/LICENSE.txt

@@ -0,0 +1,674 @@
+                    GNU GENERAL PUBLIC LICENSE
+                       Version 3, 29 June 2007
+
+ Copyright (C) 2007 Free Software Foundation, Inc. <https://fsf.org/>
+ Everyone is permitted to copy and distribute verbatim copies
+ of this license document, but changing it is not allowed.
+
+                            Preamble
+
+  The GNU General Public License is a free, copyleft license for
+software and other kinds of works.
+
+  The licenses for most software and other practical works are designed
+to take away your freedom to share and change the works.  By contrast,
+the GNU General Public License is intended to guarantee your freedom to
+share and change all versions of a program--to make sure it remains free
+software for all its users.  We, the Free Software Foundation, use the
+GNU General Public License for most of our software; it applies also to
+any other work released this way by its authors.  You can apply it to
+your programs, too.
+
+  When we speak of free software, we are referring to freedom, not
+price.  Our General Public Licenses are designed to make sure that you
+have the freedom to distribute copies of free software (and charge for
+them if you wish), that you receive source code or can get it if you
+want it, that you can change the software or use pieces of it in new
+free programs, and that you know you can do these things.
+
+  To protect your rights, we need to prevent others from denying you
+these rights or asking you to surrender the rights.  Therefore, you have
+certain responsibilities if you distribute copies of the software, or if
+you modify it: responsibilities to respect the freedom of others.
+
+  For example, if you distribute copies of such a program, whether
+gratis or for a fee, you must pass on to the recipients the same
+freedoms that you received.  You must make sure that they, too, receive
+or can get the source code.  And you must show them these terms so they
+know their rights.
+
+  Developers that use the GNU GPL protect your rights with two steps:
+(1) assert copyright on the software, and (2) offer you this License
+giving you legal permission to copy, distribute and/or modify it.
+
+  For the developers' and authors' protection, the GPL clearly explains
+that there is no warranty for this free software.  For both users' and
+authors' sake, the GPL requires that modified versions be marked as
+changed, so that their problems will not be attributed erroneously to
+authors of previous versions.
+
+  Some devices are designed to deny users access to install or run
+modified versions of the software inside them, although the manufacturer
+can do so.  This is fundamentally incompatible with the aim of
+protecting users' freedom to change the software.  The systematic
+pattern of such abuse occurs in the area of products for individuals to
+use, which is precisely where it is most unacceptable.  Therefore, we
+have designed this version of the GPL to prohibit the practice for those
+products.  If such problems arise substantially in other domains, we
+stand ready to extend this provision to those domains in future versions
+of the GPL, as needed to protect the freedom of users.
+
+  Finally, every program is threatened constantly by software patents.
+States should not allow patents to restrict development and use of
+software on general-purpose computers, but in those that do, we wish to
+avoid the special danger that patents applied to a free program could
+make it effectively proprietary.  To prevent this, the GPL assures that
+patents cannot be used to render the program non-free.
+
+  The precise terms and conditions for copying, distribution and
+modification follow.
+
+                       TERMS AND CONDITIONS
+
+  0. Definitions.
+
+  "This License" refers to version 3 of the GNU General Public License.
+
+  "Copyright" also means copyright-like laws that apply to other kinds of
+works, such as semiconductor masks.
+
+  "The Program" refers to any copyrightable work licensed under this
+License.  Each licensee is addressed as "you".  "Licensees" and
+"recipients" may be individuals or organizations.
+
+  To "modify" a work means to copy from or adapt all or part of the work
+in a fashion requiring copyright permission, other than the making of an
+exact copy.  The resulting work is called a "modified version" of the
+earlier work or a work "based on" the earlier work.
+
+  A "covered work" means either the unmodified Program or a work based
+on the Program.
+
+  To "propagate" a work means to do anything with it that, without
+permission, would make you directly or secondarily liable for
+infringement under applicable copyright law, except executing it on a
+computer or modifying a private copy.  Propagation includes copying,
+distribution (with or without modification), making available to the
+public, and in some countries other activities as well.
+
+  To "convey" a work means any kind of propagation that enables other
+parties to make or receive copies.  Mere interaction with a user through
+a computer network, with no transfer of a copy, is not conveying.
+
+  An interactive user interface displays "Appropriate Legal Notices"
+to the extent that it includes a convenient and prominently visible
+feature that (1) displays an appropriate copyright notice, and (2)
+tells the user that there is no warranty for the work (except to the
+extent that warranties are provided), that licensees may convey the
+work under this License, and how to view a copy of this License.  If
+the interface presents a list of user commands or options, such as a
+menu, a prominent item in the list meets this criterion.
+
+  1. Source Code.
+
+  The "source code" for a work means the preferred form of the work
+for making modifications to it.  "Object code" means any non-source
+form of a work.
+
+  A "Standard Interface" means an interface that either is an official
+standard defined by a recognized standards body, or, in the case of
+interfaces specified for a particular programming language, one that
+is widely used among developers working in that language.
+
+  The "System Libraries" of an executable work include anything, other
+than the work as a whole, that (a) is included in the normal form of
+packaging a Major Component, but which is not part of that Major
+Component, and (b) serves only to enable use of the work with that
+Major Component, or to implement a Standard Interface for which an
+implementation is available to the public in source code form.  A
+"Major Component", in this context, means a major essential component
+(kernel, window system, and so on) of the specific operating system
+(if any) on which the executable work runs, or a compiler used to
+produce the work, or an object code interpreter used to run it.
+
+  The "Corresponding Source" for a work in object code form means all
+the source code needed to generate, install, and (for an executable
+work) run the object code and to modify the work, including scripts to
+control those activities.  However, it does not include the work's
+System Libraries, or general-purpose tools or generally available free
+programs which are used unmodified in performing those activities but
+which are not part of the work.  For example, Corresponding Source
+includes interface definition files associated with source files for
+the work, and the source code for shared libraries and dynamically
+linked subprograms that the work is specifically designed to require,
+such as by intimate data communication or control flow between those
+subprograms and other parts of the work.
+
+  The Corresponding Source need not include anything that users
+can regenerate automatically from other parts of the Corresponding
+Source.
+
+  The Corresponding Source for a work in source code form is that
+same work.
+
+  2. Basic Permissions.
+
+  All rights granted under this License are granted for the term of
+copyright on the Program, and are irrevocable provided the stated
+conditions are met.  This License explicitly affirms your unlimited
+permission to run the unmodified Program.  The output from running a
+covered work is covered by this License only if the output, given its
+content, constitutes a covered work.  This License acknowledges your
+rights of fair use or other equivalent, as provided by copyright law.
+
+  You may make, run and propagate covered works that you do not
+convey, without conditions so long as your license otherwise remains
+in force.  You may convey covered works to others for the sole purpose
+of having them make modifications exclusively for you, or provide you
+with facilities for running those works, provided that you comply with
+the terms of this License in conveying all material for which you do
+not control copyright.  Those thus making or running the covered works
+for you must do so exclusively on your behalf, under your direction
+and control, on terms that prohibit them from making any copies of
+your copyrighted material outside their relationship with you.
+
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vendor/github.com/delucks/go-subsonic/README.md

@@ -0,0 +1,156 @@
+# go-subsonic
+
+[![GoDoc](https://godoc.org/github.com/delucks/go-subsonic?status.svg)](https://godoc.org/github.com/delucks/go-subsonic)
+
+This is an API client library for Subsonic and Subsonic-compatible music servers. It is tested on Subsonic, Airsonic, and Navidrome.
+
+# Testing
+
+Tests for this library run live against instances of Subsonic-compatible servers. A docker-compose setup comes with this repo to run instances of Airsonic and Navidrome for testing purposes. To set up the test environment and run tests against the docker containers, run `test.sh`. This test runner does the following:
+
+- Download CC-licensed sample music into `build/music`
+- Set up a location for configuration for both servers in `build/data`
+- Run `docker-compose up` to initialize both servers with the music & configuration directories as volumes
+- Wait until the servers are up
+- Run tests for both Subsonic and Navidrome
+
+The tests for Subsonic run against the demo server, so please be mindful to exclude them from all but final test runs in respect for their server load. To manually tests for only one server type, use the `-run` argument to `go test` like so:
+
+```
+# Just Airsonic
+go test . -test.v -run Airsonic
+# Subsonic and Airsonic
+go test . -test.v -run '(Sub|Air)sonic'
+# Navidrome
+go test . -test.v -run Navidrome
+```
+
+If you intend on testing with your own server, modify the top-level TestSubsonic, TestAirsonic, or TestNavidrome functions with your server's address and credentials. Please test this library with other Subsonic-compatible streaming servers, and file an issue if something is amiss!
+
+# API Support
+
+## System
+
+- [x] ping (1.0.0)
+- [x] getLicense (1.0.0)
+
+## Browsing
+
+- [x] getMusicFolders (1.0.0)
+- [x] getIndexes (1.0.0)
+- [x] getMusicDirectory
+- [x] getGenres (1.9.0)
+- [x] getArtists (1.8.0)
+- [x] getArtist (1.8.0)
+- [x] getAlbum (1.0.0)
+- [x] getSong (1.8.0)
+- [x] getArtistInfo (1.11.0)
+- [x] getArtistInfo2 (1.11.0)
+- [x] getAlbumInfo (1.14.0)
+- [x] getAlbumInfo2 (1.14.0)
+- [x] getSimilarSongs (1.11.0)
+- [x] getSimilarSongs2 (1.11.0)
+- [x] getTopSongs (1.13.0)
+
+## Album/song lists
+
+- [x] getAlbumList (1.2.0)
+- [x] getAlbumList2 (1.8.0)
+- [x] getRandomSongs (1.2.0)
+- [x] getSongsByGenre (1.9.0)
+- [x] getNowPlaying (1.0.0)
+- [x] getStarred (1.8.0)
+- [x] getStarred2 (1.8.0)
+
+## Searching
+
+- [x] search2 (1.4.0)
+- [x] search3 (1.8.0)
+
+## Playlists
+
+- [x] getPlaylists (1.0.0)
+- [x] getPlaylist (1.0.0)
+- [x] createPlaylist (1.2.0)
+- [x] updatePlaylist (1.8.0)
+- [x] deletePlaylist (1.2.0)
+
+## Media retrieval
+
+- [x] stream (1.0.0)
+- [x] download (1.0.0)
+- [x] getCoverArt (1.0.0)
+- [ ] getLyrics (1.2.0)
+- [x] getAvatar (1.8.0)
+
+## Media annotation
+
+- [x] star (1.8.0)
+- [x] unstar (1.8.0)
+- [x] setRating (1.6.0)
+- [x] scrobble (1.5.0)
+
+## User management
+
+- [x] getUser (1.3.0)
+- [x] getUsers (1.8.0)
+- [x] createUser (1.1.0)
+- [x] updateUser (1.10.1)
+- [x] deleteUser (1.3.0)
+- [x] changePassword (1.1.0)
+
+## Media library scanning
+
+- [x] getScanStatus (1.15.0)
+- [x] startScan (1.15.0)
+
+## Bookmarks
+
+- [ ] getBookmarks (1.9.0)
+- [ ] createBookmark (1.9.0)
+- [ ] deleteBookmark (1.9.0)
+- [ ] getPlayQueue (1.12.0)
+- [ ] savePlayQueue (1.12.0)
+
+## Sharing
+
+- [ ] getShares (1.6.0)
+- [ ] createShare (1.6.0)
+- [ ] updateShare (1.6.0)
+- [ ] deleteShare (1.6.0)
+
+## Podcast
+
+- [ ] getPodcasts (1.6.0)
+- [ ] getNewestPodcasts (1.13.0)
+- [ ] refreshPodcasts (1.9.0)
+- [ ] createPodcastChannel (1.9.0)
+- [ ] deletePodcastChannel (1.9.0)
+- [ ] deletePodcastEpisode (1.9.0)
+- [ ] downloadPodcastEpisode (1.9.0)
+
+## Jukebox
+
+- [ ] jukeboxControl (1.2.0)
+
+## Internet radio
+
+- [ ] getInternetRadioStations (1.9.0)
+- [ ] createInternetRadioStation (1.16.0)
+- [ ] updateInternetRadioStation (1.16.0)
+- [ ] deleteInternetRadioStation (1.16.0)
+
+## Chat
+
+- [ ] getChatMessages (1.2.0)
+- [ ] addChatMessage (1.2.0)
+
+## Out of Scope
+
+Video endpoints are currently out of scope- please file an issue if you would like support for them. The deprecated "search" endpoint is unimplemented.
+
+- [ ] hls.m3u8 (1.8.0)
+- [ ] getVideos (1.8.0)
+- [ ] getVideoInfo (1.14.0)
+- [ ] getCaptions (1.14.0)
+- [ ] search (1.0.0)

vendor/github.com/delucks/go-subsonic/annotation.go

@@ -0,0 +1,100 @@
+package subsonic
+
+import (
+	"errors"
+	"fmt"
+	"net/url"
+	"strconv"
+)
+
+// StarParameters are used to identify songs, albums, and artists (or some subset of those) at the same time.
+// subsonic.Star and subsonic.Unstar both use StarParameters to identify things to star.
+type StarParameters struct {
+	SongIDs   []string
+	AlbumIDs  []string
+	ArtistIDs []string
+}
+
+// Star adds the star annotation to songs, albums, and artists.
+func (s *Client) Star(parameters StarParameters) error {
+	params := url.Values{}
+	for _, song := range parameters.SongIDs {
+		params.Add("id", song)
+	}
+	for _, album := range parameters.AlbumIDs {
+		params.Add("albumId", album)
+	}
+	for _, artist := range parameters.ArtistIDs {
+		params.Add("artistId", artist)
+	}
+	_, err := s.getValues("star", params)
+	if err != nil {
+		return err
+	}
+	return nil
+}
+
+// Unstar removes the star annotation from songs, albums, and artists.
+func (s *Client) Unstar(parameters StarParameters) error {
+	params := url.Values{}
+	for _, song := range parameters.SongIDs {
+		params.Add("id", song)
+	}
+	for _, album := range parameters.AlbumIDs {
+		params.Add("albumId", album)
+	}
+	for _, artist := range parameters.ArtistIDs {
+		params.Add("artistId", artist)
+	}
+	_, err := s.getValues("unstar", params)
+	if err != nil {
+		return err
+	}
+	return nil
+}
+
+// SetRating sets the rating of a music file.
+func (s *Client) SetRating(id string, rating int) error {
+	if rating > 5 || rating < 0 {
+		return errors.New("Rating can only be in the range 0-5")
+	}
+	params := map[string]string{
+		"id":     id,
+		"rating": fmt.Sprintf("%d", rating),
+	}
+	_, err := s.Get("setRating", params)
+	if err != nil {
+		return err
+	}
+	return nil
+}
+
+// Scrobble submits a song to last.fm if the user has configured their credentials to do so.
+//
+// Optional Parameters:
+//   time:            (Since 1.8.0) The time (in milliseconds since 1 Jan 1970) at which the song was listened to.
+//   submission:      Whether this is a "submission" (true) or a "now playing" (false) notification. Defaults to a submission.
+func (s *Client) Scrobble(id string, parameters map[string]string) error {
+	params := map[string]string{
+		"id": id,
+	}
+	if scrobbleTime, ok := parameters["time"]; ok {
+		_, err := strconv.Atoi(scrobbleTime)
+		if err != nil {
+			return fmt.Errorf("%s is not a unix-style timestamp", scrobbleTime)
+		}
+		params["time"] = scrobbleTime
+	}
+	if submission, ok := parameters["submission"]; ok {
+		_, err := strconv.ParseBool(submission)
+		if err != nil {
+			return fmt.Errorf("%s is not boolean", submission)
+		}
+		params["submission"] = submission
+	}
+	_, err := s.Get("scrobble", params)
+	if err != nil {
+		return err
+	}
+	return nil
+}

vendor/github.com/delucks/go-subsonic/browsing.go

@@ -0,0 +1,189 @@
+package subsonic
+
+// GetMusicFolders returns all configured top-level music folders.
+func (s *Client) GetMusicFolders() ([]*MusicFolder, error) {
+	resp, err := s.Get("getMusicFolders", nil)
+	if err != nil {
+		return nil, err
+	}
+	return resp.MusicFolders.MusicFolder, nil
+}
+
+// GetIndexes returns the index of entries by letter/number.
+//
+// Optional Parameters:
+//   musicFolderId:    Only return songs in the music folder with the given ID. See getMusicFolders.
+//   ifModifiedSince:  If specified, only return a result if the artist collection has changed since the given time (in milliseconds since 1 Jan 1970).
+func (s *Client) GetIndexes(parameters map[string]string) (*Indexes, error) {
+	resp, err := s.Get("getIndexes", parameters)
+	if err != nil {
+		return nil, err
+	}
+	return resp.Indexes, nil
+}
+
+// GetMusicDirectory returns a listing of all files in a music directory. Typically used to get list of albums for an artist, or list of songs for an album.
+// The ID can be an album, song, or artist - anything considered within the directory hierarchy of Subsonic.
+func (s *Client) GetMusicDirectory(id string) (*Directory, error) {
+	resp, err := s.Get("getMusicDirectory", map[string]string{"id": id})
+	if err != nil {
+		return nil, err
+	}
+	return resp.Directory, nil
+}
+
+// GetGenres returns all genres in the server.
+func (s *Client) GetGenres() ([]*Genre, error) {
+	resp, err := s.Get("getGenres", nil)
+	if err != nil {
+		return nil, err
+	}
+	return resp.Genres.Genre, nil
+}
+
+// GetArtists returns all artists in the server.
+//
+// Optional Parameters:
+//   musicFolderId:  Only return songs in the music folder with the given ID. See getMusicFolders.
+func (s *Client) GetArtists(parameters map[string]string) (*ArtistsID3, error) {
+	resp, err := s.Get("getArtists", parameters)
+	if err != nil {
+		return nil, err
+	}
+	return resp.Artists, nil
+}
+
+// GetAlbum returns an Artist by ID.
+func (s *Client) GetArtist(id string) (*ArtistID3, error) {
+	resp, err := s.Get("getArtist", map[string]string{"id": id})
+	if err != nil {
+		return nil, err
+	}
+	return resp.Artist, nil
+}
+
+// GetAlbum returns an Album by ID.
+func (s *Client) GetAlbum(id string) (*AlbumID3, error) {
+	resp, err := s.Get("getAlbum", map[string]string{"id": id})
+	if err != nil {
+		return nil, err
+	}
+	return resp.Album, nil
+}
+
+// GetSong returns a Song by ID.
+func (s *Client) GetSong(id string) (*Child, error) {
+	resp, err := s.Get("getSong", map[string]string{"id": id})
+	if err != nil {
+		return nil, err
+	}
+	return resp.Song, nil
+}
+
+// GetArtistInfo returns biography, image links, and similar artists from last.fm.
+//
+// Optional Parameters:
+//   count:             Max number of similar artists to return.
+//   includeNotPresent: Whether to return artists that are not present in the media library.
+func (s *Client) GetArtistInfo(id string, parameters map[string]string) (*ArtistInfo, error) {
+	params := make(map[string]string)
+	params["id"] = id
+	for k, v := range parameters {
+		params[k] = v
+	}
+	resp, err := s.Get("getArtistInfo", params)
+	if err != nil {
+		return nil, err
+	}
+	return resp.ArtistInfo, nil
+}
+
+// GetArtistInfo2 returns biography, image links, and similar artists like GetArtistInfo, but using id3 tags.
+//
+// Optional Parameters:
+//   count:             Max number of similar artists to return.
+//   includeNotPresent: Whether to return artists that are not present in the media library.
+func (s *Client) GetArtistInfo2(id string, parameters map[string]string) (*ArtistInfo2, error) {
+	params := make(map[string]string)
+	params["id"] = id
+	for k, v := range parameters {
+		params[k] = v
+	}
+	resp, err := s.Get("getArtistInfo2", params)
+	if err != nil {
+		return nil, err
+	}
+	return resp.ArtistInfo2, nil
+}
+
+// GetAlbumInfo returns album notes, image data, etc using data from last.fm.
+// This accepts both album and song IDs.
+func (s *Client) GetAlbumInfo(id string) (*AlbumInfo, error) {
+	resp, err := s.Get("getAlbumInfo", map[string]string{"id": id})
+	if err != nil {
+		return nil, err
+	}
+	return resp.AlbumInfo, nil
+}
+
+// GetAlbumInfo2 returns the same data as GetAlbumInfo, but organized by id3 tag.
+// It only accepts album IDs.
+func (s *Client) GetAlbumInfo2(id string) (*AlbumInfo, error) {
+	resp, err := s.Get("getAlbumInfo2", map[string]string{"id": id})
+	if err != nil {
+		return nil, err
+	}
+	return resp.AlbumInfo, nil
+}
+
+// GetSimilarSongs finds similar songs to an album, track, or artist.
+// This is mostly used for radio features. This accepts artist, album, or song IDs.
+//
+// Optional Parameters:
+//   count: Number of songs to return
+func (s *Client) GetSimilarSongs(id string, parameters map[string]string) ([]*Child, error) {
+	params := make(map[string]string)
+	params["id"] = id
+	for k, v := range parameters {
+		params[k] = v
+	}
+	resp, err := s.Get("getSimilarSongs", params)
+	if err != nil {
+		return nil, err
+	}
+	return resp.SimilarSongs.Song, nil
+}
+
+// GetSimilarSongs2 finds similar songs like GetSimilarSongs, but using id3 tags.
+//
+// Optional Parameters:
+//   count: Number of songs to return
+func (s *Client) GetSimilarSongs2(id string, parameters map[string]string) ([]*Child, error) {
+	params := make(map[string]string)
+	params["id"] = id
+	for k, v := range parameters {
+		params[k] = v
+	}
+	resp, err := s.Get("getSimilarSongs2", params)
+	if err != nil {
+		return nil, err
+	}
+	return resp.SimilarSongs2.Song, nil
+}
+
+// GetTopSongs returns the top songs for a given artist by name.
+//
+// Optional Parameters:
+//   count: Number of songs to return
+func (s *Client) GetTopSongs(name string, parameters map[string]string) ([]*Child, error) {
+	params := make(map[string]string)
+	params["artist"] = name
+	for k, v := range parameters {
+		params[k] = v
+	}
+	resp, err := s.Get("getTopSongs", params)
+	if err != nil {
+		return nil, err
+	}
+	return resp.TopSongs.Song, nil
+}

vendor/github.com/delucks/go-subsonic/client.go

@@ -0,0 +1,171 @@
+// Package subsonic implements an API client library for Subsonic-compatible music streaming servers.
+//
+// This project handles communication with a remote *sonic server, but does not handle playback of media. The library user should be prepared to do something with the stream of audio in bytes, like decoding and playing that audio on a sound card.
+// The list of API endpoints implemented is available on the project's github page.
+//
+// The API is divided between functions with no suffix, and functions that have a "2" suffix (or "3" in the case of Search3).
+// Generally, things with "2" on the end are organized by file tags rather than folder structure. This is how you'd expect most music players to work and is recommended.
+// The variants without a suffix organize the library by directory structure; artists are a directory, albums are children of that directory, songs (subsonic.Child) are children of albums.
+// This has some disadvantages: possibly duplicating items with identical directory names, treating songs and albums in much the same fashion, and being more difficult to query consistently.
+package subsonic
+
+import (
+	"crypto/md5"
+	"encoding/xml"
+	"fmt"
+	"io"
+	"io/ioutil"
+	"log"
+	"math/rand"
+	"net/http"
+	"net/url"
+	"path"
+)
+
+const (
+	supportedApiVersion = "1.8.0"
+	libraryVersion      = "0.0.5"
+)
+
+type Client struct {
+	Client       *http.Client
+	BaseUrl      string
+	User         string
+	ClientName   string
+	PasswordAuth bool
+	password     string
+	salt         string
+	token        string
+}
+
+func generateSalt() string {
+	var corpus = []rune("abcdefghijklmnopqrstuvwxyzABCDEFGHIJKLMNOPQRSTUVWXYZ0123456789")
+	// length is minimum 6, but let's use ten to start
+	b := make([]rune, 10)
+	for i := range b {
+		b[i] = corpus[rand.Intn(len(corpus))]
+	}
+	return string(b)
+}
+
+// Authenticate authenticates the current user with a provided password. The password is salted before transmission and requires Subsonic > 1.13.0.
+func (s *Client) Authenticate(password string) error {
+	if s.PasswordAuth {
+		s.password = password
+	} else {
+		salt := generateSalt()
+		h := md5.New()
+		_, err := io.WriteString(h, password)
+		if err != nil {
+			return err
+		}
+		_, err = io.WriteString(h, salt)
+		if err != nil {
+			return err
+		}
+		s.salt = salt
+		s.token = fmt.Sprintf("%x", h.Sum(nil))
+	}
+
+	// Test authentication
+	// Don't use the s.Ping method because that always returns true as long as the servers is up.
+	resp, err := s.Get("ping", nil)
+	if err != nil {
+		return fmt.Errorf("Authentication failed: %s", err)
+	}
+
+	if resp.Error != nil {
+		return fmt.Errorf("Authentication failed: %s", resp.Error.Message)
+	}
+
+	return nil
+}
+
+// Request performs a HTTP request against the Subsonic server as the current user.
+func (s *Client) Request(method string, endpoint string, params url.Values) (*http.Response, error) {
+	baseUrl, err := url.Parse(s.BaseUrl)
+	if err != nil {
+		return nil, err
+	}
+	baseUrl.Path = path.Join(baseUrl.Path, "/rest/", endpoint)
+	req, err := http.NewRequest(method, baseUrl.String(), nil)
+	if err != nil {
+		return nil, err
+	}
+
+	q := req.URL.Query()
+	q.Add("f", "xml")
+	q.Add("v", supportedApiVersion)
+	q.Add("c", s.ClientName)
+	q.Add("u", s.User)
+	if s.PasswordAuth {
+		q.Add("p", s.password)
+	} else {
+		q.Add("t", s.token)
+		q.Add("s", s.salt)
+	}
+
+	for key, values := range params {
+		for _, val := range values {
+			q.Add(key, val)
+		}
+	}
+	req.URL.RawQuery = q.Encode()
+	//log.Printf("%s %s", method, req.URL.String())
+
+	resp, err := s.Client.Do(req)
+	if err != nil {
+		return nil, err
+	}
+	return resp, nil
+}
+
+// Get is a convenience interface to issue a GET request and parse the response body (99% of Subsonic API calls)
+func (s *Client) Get(endpoint string, params map[string]string) (*Response, error) {
+	parameters := url.Values{}
+	for k, v := range params {
+		parameters.Add(k, v)
+	}
+	return s.getValues(endpoint, parameters)
+}
+
+// getValues is a convenience interface to issue a GET request and parse the response body. It supports multiple values by way of the url.Values argument.
+func (s *Client) getValues(endpoint string, params url.Values) (*Response, error) {
+	response, err := s.Request("GET", endpoint, params)
+	if err != nil {
+		return nil, err
+	}
+	responseBody, err := ioutil.ReadAll(response.Body)
+	if err != nil {
+		return nil, err
+	}
+	parsed := Response{}
+	err = xml.Unmarshal(responseBody, &parsed)
+	if err != nil {
+		return nil, err
+	}
+	if parsed.Error != nil {
+		return nil, fmt.Errorf("Error #%d: %s\n", parsed.Error.Code, parsed.Error.Message)
+	}
+	//log.Printf("%s: %s\n", endpoint, string(responseBody))
+	return &parsed, nil
+}
+
+// Ping is used to test connectivity with the server. It returns true if the server is up.
+func (s *Client) Ping() bool {
+	_, err := s.Request("GET", "ping", nil)
+	if err != nil {
+		log.Println(err)
+		return false
+	}
+	return true
+}
+
+// GetLicense retrieves details about the software license. Subsonic requires a license after a 30-day trial, compatible applications have a perpetually valid license.
+func (s *Client) GetLicense() (*License, error) {
+	resp, err := s.Get("getLicense", nil)
+	if err != nil {
+		return nil, err
+	}
+	return resp.License, nil
+}

vendor/github.com/delucks/go-subsonic/docker-compose.yml

@@ -0,0 +1,26 @@
+version: "3"
+services:
+  navidrome:
+    image: deluan/navidrome:latest
+    user: 1000:1000 # should be owner of volumes
+    ports:
+      - "4533:4533"
+    restart: unless-stopped
+    environment:
+      # Put your config options customization here
+      ND_SCANINTERVAL: 1m
+      ND_LOGLEVEL: info
+      ND_SESSIONTIMEOUT: 24h
+      ND_BASEURL: ""
+    volumes:
+      - "./build/data:/data"
+      - "./build/music:/music:ro"
+  airsonic:
+    image: airsonic/airsonic:latest
+    user: 1000:1000
+    ports:
+      - "4040:4040"
+    restart: unless-stopped
+    volumes:
+      - "./build/data:/airsonic/data"
+      - "./build/music:/airsonic/music:ro"

vendor/github.com/delucks/go-subsonic/lists.go

@@ -0,0 +1,175 @@
+package subsonic
+
+import (
+	"errors"
+	"fmt"
+)
+
+func validateListType(input string) bool {
+	validTypes := map[string]bool{
+		"random":               true,
+		"newest":               true,
+		"highest":              true,
+		"frequent":             true,
+		"recent":               true,
+		"alphabeticalByName":   true,
+		"alphabeticalByArtist": true,
+		"starred":              true,
+		"byYear":               true,
+		"byGenre":              true,
+	}
+	_, ok := validTypes[input]
+	return ok
+}
+
+// GetAlbumList returns a list of random, newest, highest rated etc. albums. Similar to the album lists on the home page of the Subsonic web interface.
+//
+// Optional Parameters:
+//   size:           The number of albums to return. Max 500, default 10.
+//   offset:         The list offset. Useful if you for example want to page through the list of newest albums.
+//   fromYear:       The first year in the range. If fromYear > toYear a reverse chronological list is returned.
+//   toYear:         The last year in the range.
+//   genre:          The name of the genre, e.g., "Rock".
+//   musicFolderId:  (Since 1.11.0) Only return albums in the music folder with the given ID. See getMusicFolders.
+//
+// toYear and fromYear are required parameters when type == "byYear". genre is a required parameter when type == "byGenre".
+func (s *Client) GetAlbumList(listType string, parameters map[string]string) ([]*Child, error) {
+	if !validateListType(listType) {
+		return nil, fmt.Errorf("List type %s is invalid, see http://www.subsonic.org/pages/api.jsp#getAlbumList", listType)
+	}
+	if listType == "byYear" {
+		_, ok := parameters["fromYear"]
+		if !ok {
+			return nil, errors.New("Required argument fromYear was not found when using GetAlbumList byYear")
+		}
+		_, ok = parameters["toYear"]
+		if !ok {
+			return nil, errors.New("Required argument toYear was not found when using GetAlbumList byYear")
+		}
+	} else if listType == "byGenre" {
+		_, ok := parameters["genre"]
+		if !ok {
+			return nil, errors.New("Required argument genre was not found when using GetAlbumList byGenre")
+		}
+	}
+	params := make(map[string]string)
+	params["type"] = listType
+	for k, v := range parameters {
+		params[k] = v
+	}
+	resp, err := s.Get("getAlbumList", params)
+	if err != nil {
+		return nil, err
+	}
+	return resp.AlbumList.Album, nil
+}
+
+// GetAlbumList2 returns a list of albums like GetAlbumList, but organized according to id3 tags.
+//
+// Optional Parameters:
+//   size:           The number of albums to return. Max 500, default 10.
+//   offset:         The list offset. Useful if you for example want to page through the list of newest albums.
+//   fromYear:       The first year in the range. If fromYear > toYear a reverse chronological list is returned.
+//   toYear:         The last year in the range.
+//   genre:          The name of the genre, e.g., "Rock".
+//   musicFolderId:  (Since 1.11.0) Only return albums in the music folder with the given ID. See getMusicFolders.
+//
+// toYear and fromYear are required parameters when type == "byYear". genre is a required parameter when type == "byGenre".
+func (s *Client) GetAlbumList2(listType string, parameters map[string]string) ([]*AlbumID3, error) {
+	if !validateListType(listType) {
+		return nil, fmt.Errorf("List type %s is invalid, see http://www.subsonic.org/pages/api.jsp#getAlbumList", listType)
+	}
+	if listType == "byYear" {
+		_, ok := parameters["fromYear"]
+		if !ok {
+			return nil, errors.New("Required argument fromYear was not found when using GetAlbumList2 byYear")
+		}
+		_, ok = parameters["toYear"]
+		if !ok {
+			return nil, errors.New("Required argument toYear was not found when using GetAlbumList2 byYear")
+		}
+	} else if listType == "byGenre" {
+		_, ok := parameters["genre"]
+		if !ok {
+			return nil, errors.New("Required argument genre was not found when using GetAlbumList2 byGenre")
+		}
+	}
+	params := make(map[string]string)
+	params["type"] = listType
+	for k, v := range parameters {
+		params[k] = v
+	}
+	resp, err := s.Get("getAlbumList2", params)
+	if err != nil {
+		return nil, err
+	}
+	return resp.AlbumList2.Album, nil
+}
+
+// GetRandomSongs returns a randomly selected set of songs limited by the optional parameters.
+//
+// Optional Parameters:
+//   size:           The maximum number of songs to return. Max 500, default 10.
+//   genre:          Only returns songs belonging to this genre.
+//   fromYear:       Only return songs published after or in this year.
+//   toYear:         Only return songs published before or in this year.
+//   musicFolderId:  Only return songs in the music folder with the given ID. See getMusicFolders.
+func (s *Client) GetRandomSongs(parameters map[string]string) ([]*Child, error) {
+	resp, err := s.Get("getRandomSongs", parameters)
+	if err != nil {
+		return nil, err
+	}
+	return resp.RandomSongs.Song, nil
+}
+
+// GetSongsByGenre returns songs in a given genre name.
+//
+// Optional Parameters:
+//   count:          The maximum number of songs to return. Max 500, default 10.
+//   offset:         The offset. Useful if you want to page through the songs in a genre.
+//   musicFolderId:  Only return songs in the music folder with the given ID. See getMusicFolders.
+func (s *Client) GetSongsByGenre(name string, parameters map[string]string) ([]*Child, error) {
+	params := make(map[string]string)
+	params["genre"] = name
+	for k, v := range parameters {
+		params[k] = v
+	}
+	resp, err := s.Get("getSongsByGenre", params)
+	if err != nil {
+		return nil, err
+	}
+	return resp.SongsByGenre.Song, nil
+}
+
+// GetNowPlaying returns what is currently being played by all users.
+func (s *Client) GetNowPlaying() ([]*NowPlayingEntry, error) {
+	resp, err := s.Get("getNowPlaying", nil)
+	if err != nil {
+		return nil, err
+	}
+	return resp.NowPlaying.Entry, nil
+}
+
+// GetStarred returns starred albums, artists, and songs.
+//
+// Optional Parameters:
+//   musicFolderId:  Only return songs in the music folder with the given ID. See getMusicFolders.
+func (s *Client) GetStarred(parameters map[string]string) (*Starred, error) {
+	resp, err := s.Get("getStarred", parameters)
+	if err != nil {
+		return nil, err
+	}
+	return resp.Starred, nil
+}
+
+// GetStarred2 returns starred albums, artists, and songs arranged by id3 tag.
+//
+// Optional Parameters:
+//   musicFolderId:  Only return songs in the music folder with the given ID. See getMusicFolders.
+func (s *Client) GetStarred2(parameters map[string]string) (*Starred2, error) {
+	resp, err := s.Get("getStarred2", parameters)
+	if err != nil {
+		return nil, err
+	}
+	return resp.Starred2, nil
+}

vendor/github.com/delucks/go-subsonic/models.go

@@ -0,0 +1,679 @@
+package subsonic
+
+/* This file was automatically generated from the xsd schema provided by Subsonic, then manually modified.
+ *   http://www.subsonic.org/pages/inc/api/schema/subsonic-rest-api-1.16.1.xsd
+ *   xsdgen -o xml.go -pkg subsonic -ns "http://subsonic.org/restapi" subsonic-rest-api-1.16.1.xsd
+ * Changes from the original include:
+ * - Adding missing name (value of xml element) for each genre
+ * - Capitalize "ID" in struct names and add missing ID fields.
+ * - Merge *With* variants of structs.
+ */
+
+import (
+	"bytes"
+	"encoding/xml"
+	"time"
+)
+
+// AlbumID3 is an album that's organized by music file tags.
+type AlbumID3 struct {
+	ID        string    `xml:"id,attr"`                                    // Manually added
+	Song      []*Child  `xml:"http://subsonic.org/restapi song,omitempty"` // Merged from AlbumWithSongsID3
+	Name      string    `xml:"name,attr"`
+	Artist    string    `xml:"artist,attr,omitempty"`
+	ArtistID  string    `xml:"artistId,attr,omitempty"`
+	CoverArt  string    `xml:"coverArt,attr,omitempty"`
+	SongCount int       `xml:"songCount,attr"`
+	Duration  int       `xml:"duration,attr"`
+	PlayCount int64     `xml:"playCount,attr,omitempty"`
+	Created   time.Time `xml:"created,attr"`
+	Starred   time.Time `xml:"starred,attr,omitempty"`
+	Year      int       `xml:"year,attr,omitempty"`
+	Genre     string    `xml:"genre,attr,omitempty"`
+}
+
+// AlbumInfo is a collection of notes and links describing an album.
+type AlbumInfo struct {
+	Notes          string `xml:"http://subsonic.org/restapi notes,omitempty"`
+	MusicBrainzID  string `xml:"http://subsonic.org/restapi musicBrainzId,omitempty"`
+	LastFmUrl      string `xml:"http://subsonic.org/restapi lastFmUrl,omitempty"`
+	SmallImageUrl  string `xml:"http://subsonic.org/restapi smallImageUrl,omitempty"`
+	MediumImageUrl string `xml:"http://subsonic.org/restapi mediumImageUrl,omitempty"`
+	LargeImageUrl  string `xml:"http://subsonic.org/restapi largeImageUrl,omitempty"`
+}
+
+type albumList struct {
+	Album []*Child `xml:"http://subsonic.org/restapi album,omitempty"`
+}
+
+type albumList2 struct {
+	Album []*AlbumID3 `xml:"http://subsonic.org/restapi album,omitempty"`
+}
+
+// Artist is an artist from the server, organized in the folders pattern.
+type Artist struct {
+	ID             string    `xml:"id,attr"`
+	Name           string    `xml:"name,attr"`
+	ArtistImageUrl string    `xml:"artistImageUrl,attr,omitempty"`
+	Starred        time.Time `xml:"starred,attr,omitempty"`
+	UserRating     int       `xml:"userRating,attr,omitempty"`
+	AverageRating  float64   `xml:"averageRating,attr,omitempty"`
+}
+
+// ArtistID3 is an artist from the server, organized by ID3 tag.
+type ArtistID3 struct {
+	ID             string      `xml:"id,attr"`                                     // Manually added
+	Album          []*AlbumID3 `xml:"http://subsonic.org/restapi album,omitempty"` // Merged with ArtistWithAlbumsID3
+	Name           string      `xml:"name,attr"`
+	CoverArt       string      `xml:"coverArt,attr,omitempty"`
+	ArtistImageUrl string      `xml:"artistImageUrl,attr,omitempty"`
+	AlbumCount     int         `xml:"albumCount,attr"`
+	Starred        time.Time   `xml:"starred,attr,omitempty"`
+}
+
+// ArtistInfo is all auxillary information about an artist from GetArtistInfo.
+type ArtistInfo struct {
+	SimilarArtist  []*Artist `xml:"http://subsonic.org/restapi similarArtist,omitempty"`
+	Biography      string    `xml:"http://subsonic.org/restapi biography,omitempty"`
+	MusicBrainzID  string    `xml:"http://subsonic.org/restapi musicBrainzId,omitempty"`
+	LastFmUrl      string    `xml:"http://subsonic.org/restapi lastFmUrl,omitempty"`
+	SmallImageUrl  string    `xml:"http://subsonic.org/restapi smallImageUrl,omitempty"`
+	MediumImageUrl string    `xml:"http://subsonic.org/restapi mediumImageUrl,omitempty"`
+	LargeImageUrl  string    `xml:"http://subsonic.org/restapi largeImageUrl,omitempty"`
+}
+
+// ArtistInfo2 is all auxillary information about an artist from GetArtistInfo2, with similar artists organized by ID3 tags.
+type ArtistInfo2 struct {
+	SimilarArtist  []*ArtistID3 `xml:"http://subsonic.org/restapi similarArtist,omitempty"`
+	Biography      string       `xml:"http://subsonic.org/restapi biography,omitempty"`
+	MusicBrainzID  string       `xml:"http://subsonic.org/restapi musicBrainzId,omitempty"`
+	LastFmUrl      string       `xml:"http://subsonic.org/restapi lastFmUrl,omitempty"`
+	SmallImageUrl  string       `xml:"http://subsonic.org/restapi smallImageUrl,omitempty"`
+	MediumImageUrl string       `xml:"http://subsonic.org/restapi mediumImageUrl,omitempty"`
+	LargeImageUrl  string       `xml:"http://subsonic.org/restapi largeImageUrl,omitempty"`
+}
+
+// ArtistsID3 is an index of every artist on the server organized by ID3 tag, from getArtists.
+type ArtistsID3 struct {
+	Index           []*IndexID3 `xml:"http://subsonic.org/restapi index,omitempty"`
+	IgnoredArticles string      `xml:"ignoredArticles,attr"`
+}
+
+type Bookmark struct {
+	Entry    *Child    `xml:"http://subsonic.org/restapi entry"`
+	Position int64     `xml:"position,attr"`
+	Username string    `xml:"username,attr"`
+	Comment  string    `xml:"comment,attr,omitempty"`
+	Created  time.Time `xml:"created,attr"`
+	Changed  time.Time `xml:"changed,attr"`
+}
+
+func (t *Bookmark) MarshalXML(e *xml.Encoder, start xml.StartElement) error {
+	type T Bookmark
+	var layout struct {
+		*T
+		Created *xsdDateTime `xml:"created,attr"`
+		Changed *xsdDateTime `xml:"changed,attr"`
+	}
+	layout.T = (*T)(t)
+	layout.Created = (*xsdDateTime)(&layout.T.Created)
+	layout.Changed = (*xsdDateTime)(&layout.T.Changed)
+	return e.EncodeElement(layout, start)
+}
+func (t *Bookmark) UnmarshalXML(d *xml.Decoder, start xml.StartElement) error {
+	type T Bookmark
+	var overlay struct {
+		*T
+		Created *xsdDateTime `xml:"created,attr"`
+		Changed *xsdDateTime `xml:"changed,attr"`
+	}
+	overlay.T = (*T)(t)
+	overlay.Created = (*xsdDateTime)(&overlay.T.Created)
+	overlay.Changed = (*xsdDateTime)(&overlay.T.Changed)
+	return d.DecodeElement(&overlay, &start)
+}
+
+type bookmarks struct {
+	Bookmark []*Bookmark `xml:"http://subsonic.org/restapi bookmark,omitempty"`
+}
+
+type ChatMessage struct {
+	Username string `xml:"username,attr"`
+	Time     int64  `xml:"time,attr"`
+	Message  string `xml:"message,attr"`
+}
+
+type chatMessages struct {
+	ChatMessage []*ChatMessage `xml:"http://subsonic.org/restapi chatMessage,omitempty"`
+}
+
+// Child is a song, or a generic entry in the hierarchical directory structure of the database.
+// You can tell if Child is used as a song contextually based on what it was returned by, or if the IsDir boolean was set to true.
+type Child struct {
+	ID                    string    `xml:"id,attr"` // Manually added
+	Parent                string    `xml:"parent,attr,omitempty"`
+	IsDir                 bool      `xml:"isDir,attr"`
+	Title                 string    `xml:"title,attr"`
+	Album                 string    `xml:"album,attr,omitempty"`
+	Artist                string    `xml:"artist,attr,omitempty"`
+	Track                 int       `xml:"track,attr,omitempty"`
+	Year                  int       `xml:"year,attr,omitempty"`
+	Genre                 string    `xml:"genre,attr,omitempty"`
+	CoverArt              string    `xml:"coverArt,attr,omitempty"`
+	Size                  int64     `xml:"size,attr,omitempty"`
+	ContentType           string    `xml:"contentType,attr,omitempty"`
+	Suffix                string    `xml:"suffix,attr,omitempty"`
+	TranscodedContentType string    `xml:"transcodedContentType,attr,omitempty"`
+	TranscodedSuffix      string    `xml:"transcodedSuffix,attr,omitempty"`
+	Duration              int       `xml:"duration,attr,omitempty"`
+	BitRate               int       `xml:"bitRate,attr,omitempty"`
+	Path                  string    `xml:"path,attr,omitempty"`
+	IsVideo               bool      `xml:"isVideo,attr,omitempty"`
+	UserRating            int       `xml:"userRating,attr,omitempty"`
+	AverageRating         float64   `xml:"averageRating,attr,omitempty"`
+	PlayCount             int64     `xml:"playCount,attr,omitempty"`
+	DiscNumber            int       `xml:"discNumber,attr,omitempty"`
+	Created               time.Time `xml:"created,attr,omitempty"`
+	Starred               time.Time `xml:"starred,attr,omitempty"`
+	AlbumID               string    `xml:"albumId,attr,omitempty"`
+	ArtistID              string    `xml:"artistId,attr,omitempty"`
+	Type                  string    `xml:"type,attr,omitempty"` // May be one of music, podcast, audiobook, video
+	BookmarkPosition      int64     `xml:"bookmarkPosition,attr,omitempty"`
+	OriginalWidth         int       `xml:"originalWidth,attr,omitempty"`
+	OriginalHeight        int       `xml:"originalHeight,attr,omitempty"`
+}
+
+// Directory is an entry in the hierarchical folder structure organization of the server database.
+type Directory struct {
+	ID            string    `xml:"id,attr"` // Manually added
+	Child         []*Child  `xml:"http://subsonic.org/restapi child,omitempty"`
+	Parent        string    `xml:"parent,attr,omitempty"`
+	Name          string    `xml:"name,attr"`
+	Starred       time.Time `xml:"starred,attr,omitempty"`
+	UserRating    int       `xml:"userRating,attr,omitempty"`
+	AverageRating float64   `xml:"averageRating,attr,omitempty"`
+	PlayCount     int64     `xml:"playCount,attr,omitempty"`
+}
+
+type Error struct {
+	Code    int    `xml:"code,attr"`
+	Message string `xml:"message,attr,omitempty"`
+}
+
+// Genre is a style tag for a collection of songs and albums.
+type Genre struct {
+	Name       string `xml:",chardata"` // Added manually
+	SongCount  int    `xml:"songCount,attr"`
+	AlbumCount int    `xml:"albumCount,attr"`
+}
+
+type genres struct {
+	Genre []*Genre `xml:"http://subsonic.org/restapi genre,omitempty"`
+}
+
+// Index is a collection of artists that begin with the same first letter, along with that letter or category.
+type Index struct {
+	Artist []*Artist `xml:"http://subsonic.org/restapi artist,omitempty"`
+	Name   string    `xml:"name,attr"`
+}
+
+// Index is a collection of artists by ID3 tag that begin with the same first letter, along with that letter or category.
+type IndexID3 struct {
+	Artist []*ArtistID3 `xml:"http://subsonic.org/restapi artist,omitempty"`
+	Name   string       `xml:"name,attr"`
+}
+
+// Indexes is the full index of the database, returned by getIndex.
+// It contains some Index structs for each letter of the DB, plus Child entries for individual tracks.
+type Indexes struct {
+	Shortcut        []*Artist `xml:"http://subsonic.org/restapi shortcut,omitempty"`
+	Index           []*Index  `xml:"http://subsonic.org/restapi index,omitempty"`
+	Child           []*Child  `xml:"http://subsonic.org/restapi child,omitempty"`
+	LastModified    int64     `xml:"lastModified,attr"`
+	IgnoredArticles string    `xml:"ignoredArticles,attr"`
+}
+
+type InternetRadioStation struct {
+	Name        string `xml:"name,attr"`
+	StreamUrl   string `xml:"streamUrl,attr"`
+	HomePageUrl string `xml:"homePageUrl,attr,omitempty"`
+}
+
+type internetRadioStations struct {
+	InternetRadioStation []*InternetRadioStation `xml:"http://subsonic.org/restapi internetRadioStation,omitempty"`
+}
+
+type JukeboxPlaylist struct {
+	Entry        []*Child `xml:"http://subsonic.org/restapi entry,omitempty"`
+	CurrentIndex int      `xml:"currentIndex,attr"`
+	Playing      bool     `xml:"playing,attr"`
+	Gain         float32  `xml:"gain,attr"`
+	Position     int      `xml:"position,attr,omitempty"`
+}
+
+type JukeboxStatus struct {
+	CurrentIndex int     `xml:"currentIndex,attr"`
+	Playing      bool    `xml:"playing,attr"`
+	Gain         float32 `xml:"gain,attr"`
+	Position     int     `xml:"position,attr,omitempty"`
+}
+
+// License contains information about the Subsonic server's license validity and contact information in the case of a trial subscription.
+type License struct {
+	Valid          bool      `xml:"valid,attr"`
+	Email          string    `xml:"email,attr,omitempty"`
+	LicenseExpires time.Time `xml:"licenseExpires,attr,omitempty"`
+	TrialExpires   time.Time `xml:"trialExpires,attr,omitempty"`
+}
+
+type Lyrics struct {
+	Artist string `xml:"artist,attr,omitempty"`
+	Title  string `xml:"title,attr,omitempty"`
+}
+
+// MusicFolder is a representation of a source of music files added to the server. It is identified primarily by the numeric ID.
+type MusicFolder struct {
+	ID   string `xml:"id,attr"`
+	Name string `xml:"name,attr,omitempty"`
+}
+
+type musicFolders struct {
+	MusicFolder []*MusicFolder `xml:"http://subsonic.org/restapi musicFolder,omitempty"`
+}
+
+type newestPodcasts struct {
+	Episode []*PodcastEpisode `xml:"http://subsonic.org/restapi episode,omitempty"`
+}
+
+type nowPlaying struct {
+	Entry []*NowPlayingEntry `xml:"http://subsonic.org/restapi entry,omitempty"`
+}
+
+// NowPlayingEntry is one individual stream coming from the server along with information about who was streaming it.
+type NowPlayingEntry struct {
+	Username              string    `xml:"username,attr"`
+	MinutesAgo            int       `xml:"minutesAgo,attr"`
+	PlayerID              int       `xml:"playerId,attr"`
+	PlayerName            string    `xml:"playerName,attr,omitempty"`
+	Parent                string    `xml:"parent,attr,omitempty"`
+	IsDir                 bool      `xml:"isDir,attr"`
+	Title                 string    `xml:"title,attr"`
+	Album                 string    `xml:"album,attr,omitempty"`
+	Artist                string    `xml:"artist,attr,omitempty"`
+	Track                 int       `xml:"track,attr,omitempty"`
+	Year                  int       `xml:"year,attr,omitempty"`
+	Genre                 string    `xml:"genre,attr,omitempty"`
+	CoverArt              string    `xml:"coverArt,attr,omitempty"`
+	Size                  int64     `xml:"size,attr,omitempty"`
+	ContentType           string    `xml:"contentType,attr,omitempty"`
+	Suffix                string    `xml:"suffix,attr,omitempty"`
+	TranscodedContentType string    `xml:"transcodedContentType,attr,omitempty"`
+	TranscodedSuffix      string    `xml:"transcodedSuffix,attr,omitempty"`
+	Duration              int       `xml:"duration,attr,omitempty"`
+	BitRate               int       `xml:"bitRate,attr,omitempty"`
+	Path                  string    `xml:"path,attr,omitempty"`
+	IsVideo               bool      `xml:"isVideo,attr,omitempty"`
+	UserRating            int       `xml:"userRating,attr,omitempty"`
+	AverageRating         float64   `xml:"averageRating,attr,omitempty"`
+	PlayCount             int64     `xml:"playCount,attr,omitempty"`
+	DiscNumber            int       `xml:"discNumber,attr,omitempty"`
+	Created               time.Time `xml:"created,attr,omitempty"`
+	Starred               time.Time `xml:"starred,attr,omitempty"`
+	AlbumID               string    `xml:"albumId,attr,omitempty"`
+	ArtistID              string    `xml:"artistId,attr,omitempty"`
+	Type                  string    `xml:"type,attr,omitempty"` // May be one of music, podcast, audiobook, video
+	BookmarkPosition      int64     `xml:"bookmarkPosition,attr,omitempty"`
+	OriginalWidth         int       `xml:"originalWidth,attr,omitempty"`
+	OriginalHeight        int       `xml:"originalHeight,attr,omitempty"`
+}
+
+type PlayQueue struct {
+	Entry     []*Child  `xml:"http://subsonic.org/restapi entry,omitempty"`
+	Current   int       `xml:"current,attr,omitempty"`
+	Position  int64     `xml:"position,attr,omitempty"`
+	Username  string    `xml:"username,attr"`
+	Changed   time.Time `xml:"changed,attr"`
+	ChangedBy string    `xml:"changedBy,attr"`
+}
+
+func (t *PlayQueue) MarshalXML(e *xml.Encoder, start xml.StartElement) error {
+	type T PlayQueue
+	var layout struct {
+		*T
+		Changed *xsdDateTime `xml:"changed,attr"`
+	}
+	layout.T = (*T)(t)
+	layout.Changed = (*xsdDateTime)(&layout.T.Changed)
+	return e.EncodeElement(layout, start)
+}
+func (t *PlayQueue) UnmarshalXML(d *xml.Decoder, start xml.StartElement) error {
+	type T PlayQueue
+	var overlay struct {
+		*T
+		Changed *xsdDateTime `xml:"changed,attr"`
+	}
+	overlay.T = (*T)(t)
+	overlay.Changed = (*xsdDateTime)(&overlay.T.Changed)
+	return d.DecodeElement(&overlay, &start)
+}
+
+// Playlist is a collection of songs with metadata like a name, comment, and information about the total duration of the playlist.
+type Playlist struct {
+	ID          string    `xml:"id,attr"`                                     // Added manually
+	Entry       []*Child  `xml:"http://subsonic.org/restapi entry,omitempty"` // Merged from PlaylistWithSongs
+	AllowedUser []string  `xml:"http://subsonic.org/restapi allowedUser,omitempty"`
+	Name        string    `xml:"name,attr"`
+	Comment     string    `xml:"comment,attr,omitempty"`
+	Owner       string    `xml:"owner,attr,omitempty"`
+	Public      bool      `xml:"public,attr,omitempty"`
+	SongCount   int       `xml:"songCount,attr"`
+	Duration    int       `xml:"duration,attr"`
+	Created     time.Time `xml:"created,attr"`
+	Changed     time.Time `xml:"changed,attr"`
+	CoverArt    string    `xml:"coverArt,attr,omitempty"`
+}
+
+type playlists struct {
+	Playlist []*Playlist `xml:"http://subsonic.org/restapi playlist,omitempty"`
+}
+
+type PodcastChannel struct {
+	Episode          []*PodcastEpisode `xml:"http://subsonic.org/restapi episode,omitempty"`
+	Url              string            `xml:"url,attr"`
+	Title            string            `xml:"title,attr,omitempty"`
+	Description      string            `xml:"description,attr,omitempty"`
+	CoverArt         string            `xml:"coverArt,attr,omitempty"`
+	OriginalImageUrl string            `xml:"originalImageUrl,attr,omitempty"`
+	Status           string            `xml:"status,attr"` // May be one of new, downloading, completed, error, deleted, skipped
+	ErrorMessage     string            `xml:"errorMessage,attr,omitempty"`
+}
+
+type PodcastEpisode struct {
+	StreamID              string    `xml:"streamId,attr,omitempty"`
+	ChannelID             string    `xml:"channelId,attr"`
+	Description           string    `xml:"description,attr,omitempty"`
+	Status                string    `xml:"status,attr"` // May be one of new, downloading, completed, error, deleted, skipped
+	PublishDate           time.Time `xml:"publishDate,attr,omitempty"`
+	Parent                string    `xml:"parent,attr,omitempty"`
+	IsDir                 bool      `xml:"isDir,attr"`
+	Title                 string    `xml:"title,attr"`
+	Album                 string    `xml:"album,attr,omitempty"`
+	Artist                string    `xml:"artist,attr,omitempty"`
+	Track                 int       `xml:"track,attr,omitempty"`
+	Year                  int       `xml:"year,attr,omitempty"`
+	Genre                 string    `xml:"genre,attr,omitempty"`
+	CoverArt              string    `xml:"coverArt,attr,omitempty"`
+	Size                  int64     `xml:"size,attr,omitempty"`
+	ContentType           string    `xml:"contentType,attr,omitempty"`
+	Suffix                string    `xml:"suffix,attr,omitempty"`
+	TranscodedContentType string    `xml:"transcodedContentType,attr,omitempty"`
+	TranscodedSuffix      string    `xml:"transcodedSuffix,attr,omitempty"`
+	Duration              int       `xml:"duration,attr,omitempty"`
+	BitRate               int       `xml:"bitRate,attr,omitempty"`
+	Path                  string    `xml:"path,attr,omitempty"`
+	IsVideo               bool      `xml:"isVideo,attr,omitempty"`
+	UserRating            int       `xml:"userRating,attr,omitempty"`
+	AverageRating         float64   `xml:"averageRating,attr,omitempty"`
+	PlayCount             int64     `xml:"playCount,attr,omitempty"`
+	DiscNumber            int       `xml:"discNumber,attr,omitempty"`
+	Created               time.Time `xml:"created,attr,omitempty"`
+	Starred               time.Time `xml:"starred,attr,omitempty"`
+	AlbumID               string    `xml:"albumId,attr,omitempty"`
+	ArtistID              string    `xml:"artistId,attr,omitempty"`
+	Type                  string    `xml:"type,attr,omitempty"` // May be one of music, podcast, audiobook, video
+	BookmarkPosition      int64     `xml:"bookmarkPosition,attr,omitempty"`
+	OriginalWidth         int       `xml:"originalWidth,attr,omitempty"`
+	OriginalHeight        int       `xml:"originalHeight,attr,omitempty"`
+}
+
+func (t *PodcastEpisode) MarshalXML(e *xml.Encoder, start xml.StartElement) error {
+	type T PodcastEpisode
+	var layout struct {
+		*T
+		PublishDate *xsdDateTime `xml:"publishDate,attr,omitempty"`
+		Created     *xsdDateTime `xml:"created,attr,omitempty"`
+		Starred     *xsdDateTime `xml:"starred,attr,omitempty"`
+	}
+	layout.T = (*T)(t)
+	layout.PublishDate = (*xsdDateTime)(&layout.T.PublishDate)
+	layout.Created = (*xsdDateTime)(&layout.T.Created)
+	layout.Starred = (*xsdDateTime)(&layout.T.Starred)
+	return e.EncodeElement(layout, start)
+}
+func (t *PodcastEpisode) UnmarshalXML(d *xml.Decoder, start xml.StartElement) error {
+	type T PodcastEpisode
+	var overlay struct {
+		*T
+		PublishDate *xsdDateTime `xml:"publishDate,attr,omitempty"`
+		Created     *xsdDateTime `xml:"created,attr,omitempty"`
+		Starred     *xsdDateTime `xml:"starred,attr,omitempty"`
+	}
+	overlay.T = (*T)(t)
+	overlay.PublishDate = (*xsdDateTime)(&overlay.T.PublishDate)
+	overlay.Created = (*xsdDateTime)(&overlay.T.Created)
+	overlay.Starred = (*xsdDateTime)(&overlay.T.Starred)
+	return d.DecodeElement(&overlay, &start)
+}
+
+type podcasts struct {
+	Channel []*PodcastChannel `xml:"http://subsonic.org/restapi channel,omitempty"`
+}
+
+// Response is the main target for unmarshalling data from the API - everything within the "subsonic-response" key
+type Response struct {
+	License               *License               `xml:"http://subsonic.org/restapi license"`
+	MusicFolders          *musicFolders          `xml:"http://subsonic.org/restapi musicFolders"`
+	Indexes               *Indexes               `xml:"http://subsonic.org/restapi indexes"`
+	Directory             *Directory             `xml:"http://subsonic.org/restapi directory"`
+	Genres                *genres                `xml:"http://subsonic.org/restapi genres"`
+	Artists               *ArtistsID3            `xml:"http://subsonic.org/restapi artists"`
+	Artist                *ArtistID3             `xml:"http://subsonic.org/restapi artist"`
+	Album                 *AlbumID3              `xml:"http://subsonic.org/restapi album"`
+	Song                  *Child                 `xml:"http://subsonic.org/restapi song"`
+	NowPlaying            *nowPlaying            `xml:"http://subsonic.org/restapi nowPlaying"`
+	SearchResult2         *SearchResult2         `xml:"http://subsonic.org/restapi searchResult2"`
+	SearchResult3         *SearchResult3         `xml:"http://subsonic.org/restapi searchResult3"`
+	Playlists             *playlists             `xml:"http://subsonic.org/restapi playlists"`
+	Playlist              *Playlist              `xml:"http://subsonic.org/restapi playlist"`
+	JukeboxStatus         *JukeboxStatus         `xml:"http://subsonic.org/restapi jukeboxStatus"`
+	JukeboxPlaylist       *JukeboxPlaylist       `xml:"http://subsonic.org/restapi jukeboxPlaylist"`
+	Users                 *users                 `xml:"http://subsonic.org/restapi users"`
+	User                  *User                  `xml:"http://subsonic.org/restapi user"`
+	ChatMessages          *chatMessages          `xml:"http://subsonic.org/restapi chatMessages"`
+	AlbumList             *albumList             `xml:"http://subsonic.org/restapi albumList"`
+	AlbumList2            *albumList2            `xml:"http://subsonic.org/restapi albumList2"`
+	RandomSongs           *songs                 `xml:"http://subsonic.org/restapi randomSongs"`
+	SongsByGenre          *songs                 `xml:"http://subsonic.org/restapi songsByGenre"`
+	Lyrics                *Lyrics                `xml:"http://subsonic.org/restapi lyrics"`
+	Podcasts              *podcasts              `xml:"http://subsonic.org/restapi podcasts"`
+	NewestPodcasts        *newestPodcasts        `xml:"http://subsonic.org/restapi newestPodcasts"`
+	InternetRadioStations *internetRadioStations `xml:"http://subsonic.org/restapi internetRadioStations"`
+	Bookmarks             *bookmarks             `xml:"http://subsonic.org/restapi bookmarks"`
+	PlayQueue             *PlayQueue             `xml:"http://subsonic.org/restapi playQueue"`
+	Shares                *shares                `xml:"http://subsonic.org/restapi shares"`
+	Starred               *Starred               `xml:"http://subsonic.org/restapi starred"`
+	Starred2              *Starred2              `xml:"http://subsonic.org/restapi starred2"`
+	AlbumInfo             *AlbumInfo             `xml:"http://subsonic.org/restapi albumInfo"`
+	ArtistInfo            *ArtistInfo            `xml:"http://subsonic.org/restapi artistInfo"`
+	ArtistInfo2           *ArtistInfo2           `xml:"http://subsonic.org/restapi artistInfo2"`
+	SimilarSongs          *similarSongs          `xml:"http://subsonic.org/restapi similarSongs"`
+	SimilarSongs2         *similarSongs2         `xml:"http://subsonic.org/restapi similarSongs2"`
+	TopSongs              *topSongs              `xml:"http://subsonic.org/restapi topSongs"`
+	ScanStatus            *ScanStatus            `xml:"http://subsonic.org/restapi scanStatus"`
+	Error                 *Error                 `xml:"http://subsonic.org/restapi error"`
+	Status                string                 `xml:"status,attr"`  // May be one of ok, failed
+	Version               string                 `xml:"version,attr"` // Must match the pattern \d+\.\d+\.\d+
+}
+
+type ScanStatus struct {
+	Scanning bool  `xml:"scanning,attr"`
+	Count    int64 `xml:"count,attr,omitempty"`
+}
+
+// SearchResult2 is a collection of songs, albums, and artists related to a query.
+type SearchResult2 struct {
+	Artist []*Artist `xml:"http://subsonic.org/restapi artist,omitempty"`
+	Album  []*Child  `xml:"http://subsonic.org/restapi album,omitempty"`
+	Song   []*Child  `xml:"http://subsonic.org/restapi song,omitempty"`
+}
+
+// SearchResult3 is a collection of songs, albums, and artists related to a query.
+type SearchResult3 struct {
+	Artist []*ArtistID3 `xml:"http://subsonic.org/restapi artist,omitempty"`
+	Album  []*AlbumID3  `xml:"http://subsonic.org/restapi album,omitempty"`
+	Song   []*Child     `xml:"http://subsonic.org/restapi song,omitempty"`
+}
+
+type Share struct {
+	Entry       []*Child  `xml:"http://subsonic.org/restapi entry,omitempty"`
+	Url         string    `xml:"url,attr"`
+	Description string    `xml:"description,attr,omitempty"`
+	Username    string    `xml:"username,attr"`
+	Created     time.Time `xml:"created,attr"`
+	Expires     time.Time `xml:"expires,attr,omitempty"`
+	LastVisited time.Time `xml:"lastVisited,attr,omitempty"`
+	VisitCount  int       `xml:"visitCount,attr"`
+}
+
+func (t *Share) MarshalXML(e *xml.Encoder, start xml.StartElement) error {
+	type T Share
+	var layout struct {
+		*T
+		Created     *xsdDateTime `xml:"created,attr"`
+		Expires     *xsdDateTime `xml:"expires,attr,omitempty"`
+		LastVisited *xsdDateTime `xml:"lastVisited,attr,omitempty"`
+	}
+	layout.T = (*T)(t)
+	layout.Created = (*xsdDateTime)(&layout.T.Created)
+	layout.Expires = (*xsdDateTime)(&layout.T.Expires)
+	layout.LastVisited = (*xsdDateTime)(&layout.T.LastVisited)
+	return e.EncodeElement(layout, start)
+}
+func (t *Share) UnmarshalXML(d *xml.Decoder, start xml.StartElement) error {
+	type T Share
+	var overlay struct {
+		*T
+		Created     *xsdDateTime `xml:"created,attr"`
+		Expires     *xsdDateTime `xml:"expires,attr,omitempty"`
+		LastVisited *xsdDateTime `xml:"lastVisited,attr,omitempty"`
+	}
+	overlay.T = (*T)(t)
+	overlay.Created = (*xsdDateTime)(&overlay.T.Created)
+	overlay.Expires = (*xsdDateTime)(&overlay.T.Expires)
+	overlay.LastVisited = (*xsdDateTime)(&overlay.T.LastVisited)
+	return d.DecodeElement(&overlay, &start)
+}
+
+type shares struct {
+	Share []*Share `xml:"http://subsonic.org/restapi share,omitempty"`
+}
+
+type similarSongs struct {
+	Song []*Child `xml:"http://subsonic.org/restapi song,omitempty"`
+}
+
+type similarSongs2 struct {
+	Song []*Child `xml:"http://subsonic.org/restapi song,omitempty"`
+}
+
+type songs struct {
+	Song []*Child `xml:"http://subsonic.org/restapi song,omitempty"`
+}
+
+// Starred is a collection of songs, albums, and artists annotated by a user as starred.
+type Starred struct {
+	Artist []*Artist `xml:"http://subsonic.org/restapi artist,omitempty"`
+	Album  []*Child  `xml:"http://subsonic.org/restapi album,omitempty"`
+	Song   []*Child  `xml:"http://subsonic.org/restapi song,omitempty"`
+}
+
+// Starred2 is a collection of songs, albums, and artists organized by ID3 tags annotated by a user as starred.
+type Starred2 struct {
+	Artist []*ArtistID3 `xml:"http://subsonic.org/restapi artist,omitempty"`
+	Album  []*AlbumID3  `xml:"http://subsonic.org/restapi album,omitempty"`
+	Song   []*Child     `xml:"http://subsonic.org/restapi song,omitempty"`
+}
+
+type topSongs struct {
+	Song []*Child `xml:"http://subsonic.org/restapi song,omitempty"`
+}
+
+type User struct {
+	Folder              []int     `xml:"http://subsonic.org/restapi folder,omitempty"`
+	Username            string    `xml:"username,attr"`
+	Email               string    `xml:"email,attr,omitempty"`
+	ScrobblingEnabled   bool      `xml:"scrobblingEnabled,attr"`
+	MaxBitRate          int       `xml:"maxBitRate,attr,omitempty"`
+	AdminRole           bool      `xml:"adminRole,attr"`
+	SettingsRole        bool      `xml:"settingsRole,attr"`
+	DownloadRole        bool      `xml:"downloadRole,attr"`
+	UploadRole          bool      `xml:"uploadRole,attr"`
+	PlaylistRole        bool      `xml:"playlistRole,attr"`
+	CoverArtRole        bool      `xml:"coverArtRole,attr"`
+	CommentRole         bool      `xml:"commentRole,attr"`
+	PodcastRole         bool      `xml:"podcastRole,attr"`
+	StreamRole          bool      `xml:"streamRole,attr"`
+	JukeboxRole         bool      `xml:"jukeboxRole,attr"`
+	ShareRole           bool      `xml:"shareRole,attr"`
+	VideoConversionRole bool      `xml:"videoConversionRole,attr"`
+	AvatarLastChanged   time.Time `xml:"avatarLastChanged,attr,omitempty"`
+}
+
+func (t *User) MarshalXML(e *xml.Encoder, start xml.StartElement) error {
+	type T User
+	var layout struct {
+		*T
+		AvatarLastChanged *xsdDateTime `xml:"avatarLastChanged,attr,omitempty"`
+	}
+	layout.T = (*T)(t)
+	layout.AvatarLastChanged = (*xsdDateTime)(&layout.T.AvatarLastChanged)
+	return e.EncodeElement(layout, start)
+}
+func (t *User) UnmarshalXML(d *xml.Decoder, start xml.StartElement) error {
+	type T User
+	var overlay struct {
+		*T
+		AvatarLastChanged *xsdDateTime `xml:"avatarLastChanged,attr,omitempty"`
+	}
+	overlay.T = (*T)(t)
+	overlay.AvatarLastChanged = (*xsdDateTime)(&overlay.T.AvatarLastChanged)
+	return d.DecodeElement(&overlay, &start)
+}
+
+type users struct {
+	User []*User `xml:"http://subsonic.org/restapi user,omitempty"`
+}
+
+type xsdDateTime time.Time
+
+func (t *xsdDateTime) UnmarshalText(text []byte) error {
+	return _unmarshalTime(text, (*time.Time)(t), "2006-01-02T15:04:05.999999999")
+}
+func (t xsdDateTime) MarshalText() ([]byte, error) {
+	return []byte((time.Time)(t).Format("2006-01-02T15:04:05.999999999")), nil
+}
+func (t xsdDateTime) MarshalXML(e *xml.Encoder, start xml.StartElement) error {
+	if (time.Time)(t).IsZero() {
+		return nil
+	}
+	m, err := t.MarshalText()
+	if err != nil {
+		return err
+	}
+	return e.EncodeElement(m, start)
+}
+func (t xsdDateTime) MarshalXMLAttr(name xml.Name) (xml.Attr, error) {
+	if (time.Time)(t).IsZero() {
+		return xml.Attr{}, nil
+	}
+	m, err := t.MarshalText()
+	return xml.Attr{Name: name, Value: string(m)}, err
+}
+func _unmarshalTime(text []byte, t *time.Time, format string) (err error) {
+	s := string(bytes.TrimSpace(text))
+	*t, err = time.Parse(format, s)
+	if _, ok := err.(*time.ParseError); ok {
+		*t, err = time.Parse(format+"Z07:00", s)
+	}
+	return err
+}

vendor/github.com/delucks/go-subsonic/playlist.go

@@ -0,0 +1,76 @@
+package subsonic
+
+import "errors"
+
+// GetPlaylists returns all playlists a user is allowed to play.
+//
+// Optional Parameters:
+//   user: get playlists visible to this username rather than the current user. Must have admin permission.
+func (s *Client) GetPlaylists(parameters map[string]string) ([]*Playlist, error) {
+	resp, err := s.Get("getPlaylists", parameters)
+	if err != nil {
+		return nil, err
+	}
+	return resp.Playlists.Playlist, nil
+}
+
+// GetPlaylist returns a listing of files in a saved playlist.
+func (s *Client) GetPlaylist(id string) (*Playlist, error) {
+	resp, err := s.Get("getPlaylist", map[string]string{"id": id})
+	if err != nil {
+		return nil, err
+	}
+	return resp.Playlist, nil
+}
+
+// CreatePlaylist creates (or updates) a playlist.
+//
+// Optional Parameters:
+//   songId:     ID of a song in the playlist. Use one songId parameter for each song in the playlist.
+// Mutually Exclusive Parameters:
+//   playlistId: The playlist ID.
+//   name:       The human-readable name of the playlist.
+//
+// This returns a Playlist object in Subsonic > 1.14.0, so it cannot consistently return a *Playlist
+func (s *Client) CreatePlaylist(parameters map[string]string) error {
+	_, idPresent := parameters["playlistId"]
+	_, namePresent := parameters["name"]
+	if !(idPresent || namePresent) {
+		return errors.New("One of name or playlistId is mandatory, to create or update a playlist respectively")
+	}
+	_, err := s.Get("createPlaylist", parameters)
+	if err != nil {
+		return err
+	}
+	return nil
+}
+
+// UpdatePlaylist updates a playlist. Only the owner of a playlist is allowed to update it.
+//
+// Optional Parameters:
+//   name:              The human-readable name of the playlist.
+//   comment:           The playlist comment.
+//   public:            true if the playlist should be visible to all users, false otherwise.
+//   songIdToAdd:       Add this song with this ID to the playlist. Multiple parameters allowed.
+//   songIndexToRemove: Remove the song at this position in the playlist. Multiple parameters allowed.
+func (s *Client) UpdatePlaylist(playlistId string, parameters map[string]string) error {
+	params := make(map[string]string)
+	for k, v := range parameters {
+		params[k] = v
+	}
+	params["playlistId"] = playlistId
+	_, err := s.Get("updatePlaylist", params)
+	if err != nil {
+		return err
+	}
+	return nil
+}
+
+// DeletePlaylist deletes a saved playlist.
+func (s *Client) DeletePlaylist(playlistId string) error {
+	_, err := s.Get("deletePlaylist", map[string]string{"id": playlistId})
+	if err != nil {
+		return err
+	}
+	return nil
+}

vendor/github.com/delucks/go-subsonic/retrieval.go

@@ -0,0 +1,160 @@
+package subsonic
+
+import (
+	"encoding/xml"
+	"fmt"
+	"image"
+	"io"
+	"io/ioutil"
+	"net/url"
+	"strings"
+
+	_ "image/gif"
+	_ "image/jpeg"
+	_ "image/png"
+)
+
+// Stream returns the contents of a song, optionally transcoded, from the server.
+//
+// Optional Parameters:
+//   maxBitRate:             (Since 1.2.0) If specified, the server will attempt to limit the bitrate to this value, in kilobits per second. If set to zero, no limit is imposed.
+//   format:                 (Since 1.6.0) Specifies the preferred target format (e.g., "mp3" or "flv") in case there are multiple applicable transcodings.  Starting with 1.9.0 you can use the special value "raw" to disable transcoding.
+//   timeOffset:             Only applicable to video streaming. If specified, start streaming at the given offset (in seconds) into the video. Typically used to implement video skipping.
+//   size:                   (Since 1.6.0) Only applicable to video streaming. Requested video size specified as WxH, for instance "640x480".
+//   estimateContentLength:  (Since 1.8.0). If set to "true", the Content-Length HTTP header will be set to an estimated value for transcoded or downsampled media.
+//   converted:              (Since 1.14.0) Only applicable to video streaming. Subsonic can optimize videos for streaming by converting them to MP4. If a conversion exists for the video in question, then setting this parameter to "true" will cause the converted video to be returned instead of the original.
+func (s *Client) Stream(id string, parameters map[string]string) (io.Reader, error) {
+	params := url.Values{}
+	params.Add("id", id)
+	for k, v := range parameters {
+		params.Add(k, v)
+	}
+	response, err := s.Request("GET", "stream", params)
+	if err != nil {
+		return nil, err
+	}
+	contentType := response.Header.Get("Content-Type")
+	if strings.HasPrefix(contentType, "text/xml") || strings.HasPrefix(contentType, "application/xml") {
+		// An error was returned
+		responseBody, err := ioutil.ReadAll(response.Body)
+		if err != nil {
+			return nil, err
+		}
+		resp := Response{}
+		err = xml.Unmarshal(responseBody, &resp)
+		if err != nil {
+			return nil, err
+		}
+		if resp.Error != nil {
+			err = fmt.Errorf("Error #%d: %s\n", resp.Error.Code, resp.Error.Message)
+		} else {
+			err = fmt.Errorf("An error occurred: %#v\n", resp)
+		}
+		return nil, err
+	}
+	return response.Body, nil
+}
+
+// Download returns a given media file. Similar to stream, but this method returns the original media data without transcoding or downsampling.
+func (s *Client) Download(id string) (io.Reader, error) {
+	params := url.Values{}
+	params.Add("id", id)
+	response, err := s.Request("GET", "download", params)
+	if err != nil {
+		return nil, err
+	}
+	contentType := response.Header.Get("Content-Type")
+	if strings.HasPrefix(contentType, "text/xml") || strings.HasPrefix(contentType, "application/xml") {
+		// An error was returned
+		responseBody, err := ioutil.ReadAll(response.Body)
+		if err != nil {
+			return nil, err
+		}
+		resp := Response{}
+		err = xml.Unmarshal(responseBody, &resp)
+		if err != nil {
+			return nil, err
+		}
+		if resp.Error != nil {
+			err = fmt.Errorf("Error #%d: %s\n", resp.Error.Code, resp.Error.Message)
+		} else {
+			err = fmt.Errorf("An error occurred: %#v\n", resp)
+		}
+		return nil, err
+	}
+	return response.Body, nil
+}
+
+// GetCoverArt returns a cover art image for a song, album, or artist.
+//
+// Optional Parameters:
+//   size:            If specified, scale image to this size.
+func (s *Client) GetCoverArt(id string, parameters map[string]string) (image.Image, error) {
+	params := url.Values{}
+	params.Add("id", id)
+	if size, ok := parameters["size"]; ok {
+		params.Add("size", size)
+	}
+	response, err := s.Request("GET", "getCoverArt", params)
+	if err != nil {
+		return nil, err
+	}
+	contentType := response.Header.Get("Content-Type")
+	if strings.HasPrefix(contentType, "text/xml") || strings.HasPrefix(contentType, "application/xml") {
+		// An error was returned
+		responseBody, err := ioutil.ReadAll(response.Body)
+		if err != nil {
+			return nil, err
+		}
+		resp := Response{}
+		err = xml.Unmarshal(responseBody, &resp)
+		if err != nil {
+			return nil, err
+		}
+		if resp.Error != nil {
+			err = fmt.Errorf("Error #%d: %s\n", resp.Error.Code, resp.Error.Message)
+		} else {
+			err = fmt.Errorf("An error occurred: %#v\n", resp)
+		}
+		return nil, err
+	}
+	image, _, err := image.Decode(response.Body)
+	if err != nil {
+		return nil, err
+	}
+	return image, nil
+}
+
+// GetAvatar returns the avatar (personal image) for a user.
+func (s *Client) GetAvatar(username string) (image.Image, error) {
+	params := url.Values{}
+	params.Add("username", username)
+	response, err := s.Request("GET", "getAvatar", params)
+	if err != nil {
+		return nil, err
+	}
+	contentType := response.Header.Get("Content-Type")
+	if strings.HasPrefix(contentType, "text/xml") || strings.HasPrefix(contentType, "application/xml") {
+		// An error was returned
+		responseBody, err := ioutil.ReadAll(response.Body)
+		if err != nil {
+			return nil, err
+		}
+		resp := Response{}
+		err = xml.Unmarshal(responseBody, &resp)
+		if err != nil {
+			return nil, err
+		}
+		if resp.Error != nil {
+			err = fmt.Errorf("Error #%d: %s\n", resp.Error.Code, resp.Error.Message)
+		} else {
+			err = fmt.Errorf("An error occurred: %#v\n", resp)
+		}
+		return nil, err
+	}
+	image, _, err := image.Decode(response.Body)
+	if err != nil {
+		return nil, err
+	}
+	return image, nil
+}

vendor/github.com/delucks/go-subsonic/scanning.go

@@ -0,0 +1,19 @@
+package subsonic
+
+// GetScanStatus returns the current status for media library scanning.
+func (c *Client) GetScanStatus() (*ScanStatus, error) {
+	resp, err := c.Get("getScanStatus", nil)
+	if err != nil {
+		return nil, err
+	}
+	return resp.ScanStatus, nil
+}
+
+// StartScan initiates a rescan of the media libraries.
+func (c *Client) StartScan() (*ScanStatus, error) {
+	resp, err := c.Get("startScan", nil)
+	if err != nil {
+		return nil, err
+	}
+	return resp.ScanStatus, nil
+}

vendor/github.com/delucks/go-subsonic/search.go

@@ -0,0 +1,46 @@
+package subsonic
+
+// Search2 returns albums, artists and songs matching the given search criteria. Supports paging through the result.
+//
+// Optional Parameters:
+//   artistCount:     Maximum number of artists to return. (Default 20)
+//   artistOffset:    Search result offset for artists. Used for paging.
+//   albumCount:      Maximum number of albums to return. (Default 20)
+//   albumOffset:     Search result offset for albums. Used for paging.
+//   songCount:       Maximum number of songs to return. (Default 20)
+//   songOffset:      Search result offset for songs. Used for paging.
+//   musicFolderId:   (Since 1.12.0) Only return results from the music folder with the given ID. See getMusicFolders.
+func (s *Client) Search2(query string, parameters map[string]string) (*SearchResult2, error) {
+	params := make(map[string]string)
+	params["query"] = query
+	for k, v := range parameters {
+		params[k] = v
+	}
+	resp, err := s.Get("search2", params)
+	if err != nil {
+		return nil, err
+	}
+	return resp.SearchResult2, nil
+}
+
+// Search3 returns albums, artists and songs matching the given search criteria like Search2, but organized according to id3 tags.
+// Optional Parameters:
+//   artistCount:     Maximum number of artists to return. (Default 20)
+//   artistOffset:    Search result offset for artists. Used for paging.
+//   albumCount:      Maximum number of albums to return. (Default 20)
+//   albumOffset:     Search result offset for albums. Used for paging.
+//   songCount:       Maximum number of songs to return. (Default 20)
+//   songOffset:      Search result offset for songs. Used for paging.
+//   musicFolderId:   (Since 1.12.0) Only return results from the music folder with the given ID. See getMusicFolders.
+func (s *Client) Search3(query string, parameters map[string]string) (*SearchResult3, error) {
+	params := make(map[string]string)
+	params["query"] = query
+	for k, v := range parameters {
+		params[k] = v
+	}
+	resp, err := s.Get("search3", params)
+	if err != nil {
+		return nil, err
+	}
+	return resp.SearchResult3, nil
+}

vendor/github.com/delucks/go-subsonic/test.sh

@@ -0,0 +1,133 @@
+#!/usr/bin/env bash
+
+# This script performs automated testing of go-subsonic on Airsonic and Navidrome music servers.
+# It wraps docker-compose to download a sample music library before creating instances of these servers,
+# allowing the go-subsonic tests to run fully against the sample instances.
+
+err() {
+  echo "$1" >&2
+  exit 1
+}
+
+log() {
+  SEV="$1"
+  shift
+  echo "$(date +%Y-%m-%d\ %T) - ${SEV^^} - $*" >&2
+}
+
+for dependency in curl docker-compose; do
+  hash "$dependency" 2>/dev/null || err "$dependency must be installed"
+done
+
+SOURCE_DIR="$( cd "$( dirname "${BASH_SOURCE[0]}" )" >/dev/null 2>&1 && pwd )"
+NAVIDROME_PORT=4533
+
+download_audionautix() {
+  # Audionautix Acoustic is a CC licensed album from Jason Shaw available from Free Music Archive.
+  # https://archive.org/details/Audionautix_Acoustic-9870
+  BASEURL="https://archive.org/download/Audionautix_Acoustic-9870/Jason_Shaw_-_"
+  TRACKS=("12_MORNINGS.ogg" "ACOUSTIC_BLUES.ogg" "FUNKY_JUNKY.ogg" "JENNYS_THEME.ogg" "LAZY_DAY.ogg" "MOUNTAIN_SUN.ogg" "ONE_FINE_DAY.ogg" "RIVER_MEDITATION.ogg" "ROCKY_TOP.ogg" "RUNNING_WATERS.ogg" "SERENITY.ogg" "SIDEWALK.ogg" "SNAPPY.ogg" "SOLO_ACOUSTIC_GUITAR.ogg" "SOUTH_OF_THE_BORDER.ogg" "TENNESEE_HAYRIDE.ogg" "THINGAMAJIG.ogg" "TRAVEL_LIGHT.ogg" "WHEELS.ogg" "WORDS.ogg")
+  DESTINATION="${SOURCE_DIR}/build/music/Jason Shaw/Audionautix"
+  mkdir -p "$DESTINATION"
+  for track in "${TRACKS[@]}"
+  do
+    if test -f "$DESTINATION/$track"
+    then
+      log info "Skipping download of Audionautix $track"
+    else
+      curl -L -o "$DESTINATION/$track" "${BASEURL}${track}"
+    fi
+  done
+}
+
+download_grabbag() {
+  # Grab Bag is a CC-licensed jazz album from Jahzzar available from Free Music Archive.
+  # https://archive.org/details/Grab_Bag-12446
+  BASEURL="https://archive.org/download/Grab_Bag-12446/"
+  DESTINATION="${SOURCE_DIR}/build/music/Jahzzar/Grab Bag"
+  TRACKS=("01_-_Dummy.ogg" "02_-_Candlelight.ogg" "03_-_Trust.ogg" "04_-_Guilty.ogg" "05_-_Storm.ogg")
+  mkdir -p "$DESTINATION"
+  for track in "${TRACKS[@]}"
+  do
+    if test -f "$DESTINATION/$track"
+    then
+      log info "Skipping download of Grab Bag $track"
+    else
+      curl -L -o "$DESTINATION/$track" "${BASEURL}${track}"
+    fi
+  done
+}
+
+
+download_fourseasons() {
+  # This is a CC-licensed recording of Vivaldi's The Four Seasons performed by John Harrison with the Wichita State University Chamber Players
+  # https://archive.org/details/The_Four_Seasons_Vivaldi-10361
+  BASEURL="https://archive.org/download/The_Four_Seasons_Vivaldi-10361/John_Harrison_with_the_Wichita_State_University_Chamber_Players_-_"
+  DESTINATION="${SOURCE_DIR}/build/music/Vivaldi/The Four Seasons"
+  TRACKS=("01_-_Spring_Mvt_1_Allegro.ogg" "02_-_Spring_Mvt_2_Largo.ogg" "03_-_Spring_Mvt_3_Allegro_pastorale.ogg" "04_-_Summer_Mvt_1_Allegro_non_molto.ogg" "05_-_Summer_Mvt_2_Adagio.ogg" "06_-_Summer_Mvt_3_Presto.ogg" "07_-_Autumn_Mvt_1_Allegro.ogg" "08_-_Autumn_Mvt_2_Adagio_molto.ogg" "09_-_Autumn_Mvt_3_Allegro.ogg" "10_-_Winter_Mvt_1_Allegro_non_molto.ogg" "11_-_Winter_Mvt_2_Largo.ogg" "12_-_Winter_Mvt_3_Allegro.ogg")
+  mkdir -p "$DESTINATION"
+  for track in "${TRACKS[@]}"
+  do
+    if test -f "$DESTINATION/$track"
+    then
+      log info "Skipping download of The Four Seasons $track"
+    else
+      curl -L -o "$DESTINATION/$track" "${BASEURL}${track}"
+    fi
+  done
+}
+
+download_sample_audio() {
+  download_audionautix
+  download_grabbag
+  download_fourseasons
+}
+
+configure_airsonic() {
+  cat << DOG > build/data/airsonic.properties
+JWTKey=q7q8u331n25gkvgjiehutl3e4u
+SettingsChanged=$(date +%s)000
+# Try to force immediate library scan
+LastScanned=0
+IndexCreationInterval=1
+# We want to use id3 tags for most tests
+OrganizeByFolderStructure=false
+DOG
+}
+
+create_navidrome_user() {
+  # navidrome does not ship with a built-in user, so we call the API to create a known user before test execution
+  curl -X POST -H "Content-Type: application/json" "http://localhost:${NAVIDROME_PORT}/auth/createAdmin" --data '{"username":"admin", "password":"admin"}'
+  echo
+}
+
+clear_data_dir() {
+  rm -rf ./build/data/*
+}
+
+main() {
+  log info "Downloading sample music into ./build/music"
+  download_sample_audio
+  # Create or restart the docker containers of Airsonic and Navidrome
+  if [[ $(docker-compose top) ]]
+  then
+    # If the current composition is running, restart it to pick up possible changes
+    log warn "Downing currently running docker containers"
+    docker-compose down
+  fi
+  log info "Removing excess data"
+  clear_data_dir
+  log info "Configuring Airsonic"
+  configure_airsonic  # This must occur in the middle so settings aren't overwritten
+  log info "Bringing up containers"
+  docker-compose up -d
+  sleep 10
+  log info "Creating Navidrome administrator (admin/admin)"
+  create_navidrome_user
+  go test . -test.v -run 'Navidrome' -count=1
+  log info "Waiting 30 seconds total for Airsonic to scan the music library..."
+  sleep 20
+  go test . -test.v -run 'Airsonic' -count=1
+}
+
+main

vendor/github.com/delucks/go-subsonic/user.go

@@ -0,0 +1,104 @@
+package subsonic
+
+// GetUser gets details about a given user, including which authorization roles and folder access it has. Can be used to enable/disable certain features in the client, such as jukebox control.
+func (c *Client) GetUser(username string) (*User, error) {
+	resp, err := c.Get("getUser", map[string]string{"username": username})
+	if err != nil {
+		return nil, err
+	}
+	return resp.User, nil
+}
+
+// GetUsers gets details about all users, including which authorization roles and folder access they have. Only users with admin privileges are allowed to call this method.
+func (c *Client) GetUsers() ([]*User, error) {
+	resp, err := c.Get("getUsers", nil)
+	if err != nil {
+		return nil, err
+	}
+	return resp.Users.User, nil
+}
+
+// CreateUser creates a new Subsonic user.
+//
+// Optional Parameters		Default	Description
+//   ldapAuthenticated		false		Whether the user is authenicated in LDAP.
+//   adminRole						false		Whether the user is administrator.
+//   settingsRole					true		Whether the user is allowed to change personal settings and password.
+//   streamRole						true		Whether the user is allowed to play files.
+//   jukeboxRole					false		Whether the user is allowed to play files in jukebox mode.
+//   downloadRole					false		Whether the user is allowed to download files.
+//   uploadRole						false		Whether the user is allowed to upload files.
+//   playlistRole					false		Whether the user is allowed to create and delete playlists. Since 1.8.0, changing this role has no effect.
+//   coverArtRole					false		Whether the user is allowed to change cover art and tags.
+//   commentRole					false		Whether the user is allowed to create and edit comments and ratings.
+//   podcastRole					false		Whether the user is allowed to administrate Podcasts.
+//   shareRole						false		(Since 1.8.0) Whether the user is allowed to share files with anyone.
+//   videoConversionRole	false		(Since 1.15.0) Whether the user is allowed to start video conversions.
+//   musicFolderId				All 		(Since 1.12.0) IDs of the music folders the user is allowed access to. Include the parameter once for each folder.
+func (c *Client) CreateUser(username, password, email string, parameters map[string]string) error {
+	params := make(map[string]string)
+	params["username"] = username
+	params["password"] = password
+	params["email"] = email
+	for k, v := range parameters {
+		params[k] = v
+	}
+	_, err := c.Get("createUser", params)
+	if err != nil {
+		return err
+	}
+	return nil
+}
+
+// UpdateUser modifies an existing Subsonic user.
+//
+// Optional Parameters:
+//   password							The password of the user, either in clear text of hex-encoded.
+//   email								The email address of the user.
+//   ldapAuthenticated		Whether the user is authenicated in LDAP.
+//   adminRole						Whether the user is administrator.
+//   settingsRole					Whether the user is allowed to change personal settings and password.
+//   streamRole						Whether the user is allowed to play files.
+//   jukeboxRole					Whether the user is allowed to play files in jukebox mode.
+//   downloadRole					Whether the user is allowed to download files.
+//   uploadRole						Whether the user is allowed to upload files.
+//   coverArtRole					Whether the user is allowed to change cover art and tags.
+//   commentRole					Whether the user is allowed to create and edit comments and ratings.
+//   podcastRole					Whether the user is allowed to administrate Podcasts.
+//   shareRole						(Since 1.8.0) Whether the user is allowed to share files with anyone.
+//   videoConversionRole	(Since 1.15.0) Whether the user is allowed to start video conversions.
+//   musicFolderId				(Since 1.12.0) IDs of the music folders the user is allowed access to. Include the parameter once for each folder.
+//   maxBitRate						(Since 1.13.0) The maximum bit rate (in Kbps) for the user. Audio streams of higher bit rates are automatically downsampled to this bit rate. Legal values: 0 (no limit), 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320.
+func (c *Client) UpdateUser(username string, parameters map[string]string) error {
+	params := make(map[string]string)
+	params["username"] = username
+	for k, v := range parameters {
+		params[k] = v
+	}
+	_, err := c.Get("updateUser", params)
+	if err != nil {
+		return err
+	}
+	return nil
+}
+
+// DeleteUser deletes an existing Subsonic user.
+func (s *Client) DeleteUser(username string) error {
+	_, err := s.Get("deleteUser", map[string]string{"username": username})
+	if err != nil {
+		return err
+	}
+	return nil
+}
+
+// ChangePassword changes the password of an existing Subsonic user, using the following parameters. You can only change your own password unless you have admin privileges.
+func (c *Client) ChangePassword(username, password string) error {
+	_, err := c.Get("changePassword", map[string]string{
+		"username": username,
+		"password": password,
+	})
+	if err != nil {
+		return err
+	}
+	return nil
+}

vendor/github.com/ebitengine/oto/v3/.clang-format

@@ -0,0 +1 @@
+CommentPragmas: '^go:build'

vendor/github.com/ebitengine/oto/v3/.gitattributes

@@ -0,0 +1 @@
+internal/oboe/** linguist-vendored

vendor/github.com/ebitengine/oto/v3/.gitignore

@@ -0,0 +1,2 @@
+.DS_Store
+*~

vendor/github.com/ebitengine/oto/v3/LICENSE

@@ -0,0 +1,201 @@
+                                 Apache License
+                           Version 2.0, January 2004
+                        http://www.apache.org/licenses/
+
+   TERMS AND CONDITIONS FOR USE, REPRODUCTION, AND DISTRIBUTION
+
+   1. Definitions.
+
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+      and distribution as defined by Sections 1 through 9 of this document.
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+
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+
+      "You" (or "Your") shall mean an individual or Legal Entity
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+      any Contribution intentionally submitted for inclusion in the Work
+      by You to the Licensor shall be under the terms and conditions of
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+      the terms of any separate license agreement you may have executed
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+   6. Trademarks. This License does not grant permission to use the trade
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+
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+      Contributor provides its Contributions) on an "AS IS" BASIS,
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+       http://www.apache.org/licenses/LICENSE-2.0
+
+   Unless required by applicable law or agreed to in writing, software
+   distributed under the License is distributed on an "AS IS" BASIS,
+   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+   See the License for the specific language governing permissions and
+   limitations under the License.

vendor/github.com/ebitengine/oto/v3/README.md

@@ -0,0 +1,234 @@
+# Oto (v3)
+
+[![Go Reference](https://pkg.go.dev/badge/github.com/ebitengine/oto/v3.svg)](https://pkg.go.dev/github.com/ebitengine/oto/v3)
+[![Build Status](https://github.com/ebitengine/oto/actions/workflows/test.yml/badge.svg)](https://github.com/ebitengine/oto/actions?query=workflow%3Atest)
+
+A low-level library to play sound.
+
+- [Oto (v3)](#oto-v3)
+  - [Platforms](#platforms)
+  - [Prerequisite](#prerequisite)
+    - [macOS](#macos)
+    - [iOS](#ios)
+    - [Linux](#linux)
+    - [FreeBSD, OpenBSD](#freebsd-openbsd)
+  - [Usage](#usage)
+    - [Playing sounds from memory](#playing-sounds-from-memory)
+    - [Playing sounds by file streaming](#playing-sounds-by-file-streaming)
+    - [Advanced usage](#advanced-usage)
+  - [Crosscompiling](#crosscompiling)
+
+## Platforms
+
+- Windows (no Cgo required!)
+- macOS (no Cgo required!)
+- Linux
+- FreeBSD
+- OpenBSD
+- Android
+- iOS
+- WebAssembly
+- Nintendo Switch
+- Xbox
+
+## Prerequisite
+
+On some platforms you will need a C/C++ compiler in your path that Go can use.
+
+- iOS: On newer macOS versions type `clang` on your terminal and a dialog with installation instructions will appear if you don't have it
+  - If you get an error with clang use xcode instead `xcode-select --install`
+- Linux and other Unix systems: Should be installed by default, but if not try [GCC](https://gcc.gnu.org/) or [Clang](https://releases.llvm.org/download.html)
+
+### macOS
+
+Oto requires `AudioToolbox.framework`, but this is automatically linked.
+
+### iOS
+
+Oto requires these frameworks:
+
+- `AVFoundation.framework`
+- `AudioToolbox.framework`
+
+Add them to "Linked Frameworks and Libraries" on your Xcode project.
+
+### Linux
+
+ALSA is required. On Ubuntu or Debian, run this command:
+
+```sh
+apt install libasound2-dev
+```
+
+On RedHat-based linux distributions, run:
+
+```sh
+dnf install alsa-lib-devel
+```
+
+In most cases this command must be run by root user or through `sudo` command.
+
+### FreeBSD, OpenBSD
+
+BSD systems are not tested well. If ALSA works, Oto should work.
+
+## Usage
+
+The two main components of Oto are a `Context` and `Players`. The context handles interactions with
+the OS and audio drivers, and as such there can only be **one** context in your program.
+
+From a context you can create any number of different players, where each player is given an `io.Reader` that
+it reads bytes representing sounds from and plays.
+
+Note that a single `io.Reader` must **not** be used by multiple players.
+
+### Playing sounds from memory
+
+The following is an example of loading and playing an MP3 file:
+
+```go
+package main
+
+import (
+    "time"
+    "os"
+
+    "github.com/ebitengine/oto/v3"
+    "github.com/hajimehoshi/go-mp3"
+)
+
+func main() {
+    // Read the mp3 file into memory
+    fileBytes, err := os.ReadFile("./my-file.mp3")
+    if err != nil {
+        panic("reading my-file.mp3 failed: " + err.Error())
+    }
+
+    // Convert the pure bytes into a reader object that can be used with the mp3 decoder
+    fileBytesReader := bytes.NewReader(fileBytes)
+
+    // Decode file
+    decodedMp3, err := mp3.NewDecoder(fileBytesReader)
+    if err != nil {
+        panic("mp3.NewDecoder failed: " + err.Error())
+    }
+
+    // Prepare an Oto context (this will use your default audio device) that will
+    // play all our sounds. Its configuration can't be changed later.
+
+    op := &oto.NewContextOptions{}
+
+    // Usually 44100 or 48000. Other values might cause distortions in Oto
+    op.SampleRate = 44100
+
+    // Number of channels (aka locations) to play sounds from. Either 1 or 2.
+    // 1 is mono sound, and 2 is stereo (most speakers are stereo). 
+    op.ChannelCount = 2
+
+    // Format of the source. go-mp3's format is signed 16bit integers.
+    op.Format = oto.FormatSignedInt16LE
+
+    // Remember that you should **not** create more than one context
+    otoCtx, readyChan, err := oto.NewContext(op)
+    if err != nil {
+        panic("oto.NewContext failed: " + err.Error())
+    }
+    // It might take a bit for the hardware audio devices to be ready, so we wait on the channel.
+    <-readyChan
+
+    // Create a new 'player' that will handle our sound. Paused by default.
+    player := otoCtx.NewPlayer(decodedMp3)
+    
+    // Play starts playing the sound and returns without waiting for it (Play() is async).
+    player.Play()
+
+    // We can wait for the sound to finish playing using something like this
+    for player.IsPlaying() {
+        time.Sleep(time.Millisecond)
+    }
+
+    // Now that the sound finished playing, we can restart from the beginning (or go to any location in the sound) using seek
+    // newPos, err := player.(io.Seeker).Seek(0, io.SeekStart)
+    // if err != nil{
+    //     panic("player.Seek failed: " + err.Error())
+    // }
+    // println("Player is now at position:", newPos)
+    // player.Play()
+
+    // If you don't want the player/sound anymore simply close
+    err = player.Close()
+    if err != nil {
+        panic("player.Close failed: " + err.Error())
+    }
+}
+```
+
+### Playing sounds by file streaming
+
+The above example loads the entire file into memory and then plays it. This is great for smaller files
+but might be an issue if you are playing a long song since it would take too much memory and too long to load.
+
+In such cases you might want to stream the file. Luckily this is very simple, just use `os.Open`:
+
+```go
+package main
+
+import (
+    "bytes"
+    "os"
+    "time"
+
+    "github.com/hajimehoshi/go-mp3"
+    "github.com/hajimehoshi/oto/v3"
+)
+
+func main() {
+    // Open the file for reading. Do NOT close before you finish playing!
+    file, err := os.Open("./my-file.mp3")
+    if err != nil {
+        panic("opening my-file.mp3 failed: " + err.Error())
+    }
+
+    // Decode file. This process is done as the file plays so it won't
+    // load the whole thing into memory.
+    decodedMp3, err := mp3.NewDecoder(file)
+    if err != nil {
+        panic("mp3.NewDecoder failed: " + err.Error())
+    }
+
+    // Rest is the same...
+
+    // Close file only after you finish playing
+    file.Close()
+}
+```
+
+The only thing to note about streaming is that the *file* object must be kept alive, otherwise
+you might just play static.
+
+To keep it alive not only must you be careful about when you close it, but you might need to keep a reference
+to the original file object alive (by for example keeping it in a struct).
+
+### Advanced usage
+
+Players have their own internal audio data buffer, so while for example 200 bytes have been read from the `io.Reader` that
+doesn't mean they were all played from the audio device.
+
+Data is moved from io.Reader->internal buffer->audio device, and when the internal buffer moves data to the audio device
+is not guaranteed, so there might be a small delay. The amount of data in the buffer can be retrieved
+using `Player.UnplayedBufferSize()`.
+
+The size of the underlying buffer of a player can also be set by type-asserting the player object:
+
+```go
+myPlayer.(oto.BufferSizeSetter).SetBufferSize(newBufferSize)
+```
+
+This works because players implement a `Player` interface and a `BufferSizeSetter` interface.
+
+## Crosscompiling
+
+Crosscompiling to macOS or Windows is as easy as setting `GOOS=darwin` or `GOOS=windows`, respectively.
+
+To crosscompile for other platforms, make sure the libraries for the target architecture are installed, and set 
+`CGO_ENABLED=1` as Go disables [Cgo](https://golang.org/cmd/cgo/#hdr-Using_cgo_with_the_go_command) on crosscompiles by default.

vendor/github.com/ebitengine/oto/v3/api_darwin.go

@@ -0,0 +1,95 @@
+// Copyright 2022 The Oto Authors
+//
+// Licensed under the Apache License, Version 2.0 (the "License");
+// you may not use this file except in compliance with the License.
+// You may obtain a copy of the License at
+//
+//     http://www.apache.org/licenses/LICENSE-2.0
+//
+// Unless required by applicable law or agreed to in writing, software
+// distributed under the License is distributed on an "AS IS" BASIS,
+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+// See the License for the specific language governing permissions and
+// limitations under the License.
+
+package oto
+
+import (
+	"unsafe"
+
+	"github.com/ebitengine/purego"
+)
+
+const (
+	avAudioSessionErrorCodeCannotStartPlaying    = 0x21706c61 // '!pla'
+	avAudioSessionErrorCodeCannotInterruptOthers = 0x21696e74 // '!int'
+	avAudioSessionErrorCodeSiriIsRecording       = 0x73697269 // 'siri'
+)
+
+const (
+	kAudioFormatLinearPCM = 0x6C70636D //'lpcm'
+)
+
+const (
+	kAudioFormatFlagIsFloat = 1 << 0 // 0x1
+)
+
+type _AudioStreamBasicDescription struct {
+	mSampleRate       float64
+	mFormatID         uint32
+	mFormatFlags      uint32
+	mBytesPerPacket   uint32
+	mFramesPerPacket  uint32
+	mBytesPerFrame    uint32
+	mChannelsPerFrame uint32
+	mBitsPerChannel   uint32
+	mReserved         uint32
+}
+
+type _AudioQueueRef uintptr
+
+type _AudioTimeStamp uintptr
+
+type _AudioStreamPacketDescription struct {
+	mStartOffset            int64
+	mVariableFramesInPacket uint32
+	mDataByteSize           uint32
+}
+
+type _AudioQueueBufferRef *_AudioQueueBuffer
+
+type _AudioQueueBuffer struct {
+	mAudioDataBytesCapacity uint32
+	mAudioData              uintptr // void*
+	mAudioDataByteSize      uint32
+	mUserData               uintptr // void*
+
+	mPacketDescriptionCapacity uint32
+	mPacketDescriptions        *_AudioStreamPacketDescription
+	mPacketDescriptionCount    uint32
+}
+
+type _AudioQueueOutputCallback func(inUserData unsafe.Pointer, inAQ _AudioQueueRef, inBuffer _AudioQueueBufferRef)
+
+func initializeAPI() error {
+	toolbox, err := purego.Dlopen("/System/Library/Frameworks/AudioToolbox.framework/AudioToolbox", purego.RTLD_LAZY|purego.RTLD_GLOBAL)
+	if err != nil {
+		return err
+	}
+	purego.RegisterLibFunc(&_AudioQueueNewOutput, toolbox, "AudioQueueNewOutput")
+	purego.RegisterLibFunc(&_AudioQueueAllocateBuffer, toolbox, "AudioQueueAllocateBuffer")
+	purego.RegisterLibFunc(&_AudioQueueEnqueueBuffer, toolbox, "AudioQueueEnqueueBuffer")
+	purego.RegisterLibFunc(&_AudioQueueStart, toolbox, "AudioQueueStart")
+	purego.RegisterLibFunc(&_AudioQueuePause, toolbox, "AudioQueuePause")
+	return nil
+}
+
+var _AudioQueueNewOutput func(inFormat *_AudioStreamBasicDescription, inCallbackProc _AudioQueueOutputCallback, inUserData unsafe.Pointer, inCallbackRunLoop uintptr, inCallbackRunLoopMod uintptr, inFlags uint32, outAQ *_AudioQueueRef) uintptr
+
+var _AudioQueueAllocateBuffer func(inAQ _AudioQueueRef, inBufferByteSize uint32, outBuffer *_AudioQueueBufferRef) uintptr
+
+var _AudioQueueEnqueueBuffer func(inAQ _AudioQueueRef, inBuffer _AudioQueueBufferRef, inNumPacketDescs uint32, inPackets []_AudioStreamPacketDescription) uintptr
+
+var _AudioQueueStart func(inAQ _AudioQueueRef, inStartTime *_AudioTimeStamp) uintptr
+
+var _AudioQueuePause func(inAQ _AudioQueueRef) uintptr

vendor/github.com/ebitengine/oto/v3/api_wasapi_windows.go

@@ -0,0 +1,479 @@
+// Copyright 2022 The Oto Authors
+//
+// Licensed under the Apache License, Version 2.0 (the "License");
+// you may not use this file except in compliance with the License.
+// You may obtain a copy of the License at
+//
+//     http://www.apache.org/licenses/LICENSE-2.0
+//
+// Unless required by applicable law or agreed to in writing, software
+// distributed under the License is distributed on an "AS IS" BASIS,
+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+// See the License for the specific language governing permissions and
+// limitations under the License.
+
+package oto
+
+import (
+	"fmt"
+	"runtime"
+	"syscall"
+	"unsafe"
+
+	"golang.org/x/sys/windows"
+)
+
+var (
+	ole32 = windows.NewLazySystemDLL("ole32")
+)
+
+var (
+	procCoCreateInstance = ole32.NewProc("CoCreateInstance")
+)
+
+type _REFERENCE_TIME int64
+
+var (
+	uuidIAudioClient2       = windows.GUID{0x726778cd, 0xf60a, 0x4eda, [...]byte{0x82, 0xde, 0xe4, 0x76, 0x10, 0xcd, 0x78, 0xaa}}
+	uuidIAudioRenderClient  = windows.GUID{0xf294acfc, 0x3146, 0x4483, [...]byte{0xa7, 0xbf, 0xad, 0xdc, 0xa7, 0xc2, 0x60, 0xe2}}
+	uuidIMMDeviceEnumerator = windows.GUID{0xa95664d2, 0x9614, 0x4f35, [...]byte{0xa7, 0x46, 0xde, 0x8d, 0xb6, 0x36, 0x17, 0xe6}}
+	uuidMMDeviceEnumerator  = windows.GUID{0xbcde0395, 0xe52f, 0x467c, [...]byte{0x8e, 0x3d, 0xc4, 0x57, 0x92, 0x91, 0x69, 0x2e}}
+)
+
+const (
+	_AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM = 0x80000000
+	_AUDCLNT_STREAMFLAGS_EVENTCALLBACK  = 0x00040000
+	_AUDCLNT_STREAMFLAGS_NOPERSIST      = 0x00080000
+	_COINIT_APARTMENTTHREADED           = 0x2
+	_COINIT_MULTITHREADED               = 0
+	_REFTIMES_PER_SEC                   = 10000000
+	_SPEAKER_FRONT_CENTER               = 0x4
+	_SPEAKER_FRONT_LEFT                 = 0x1
+	_SPEAKER_FRONT_RIGHT                = 0x2
+	_WAVE_FORMAT_EXTENSIBLE             = 0xfffe
+)
+
+var (
+	_KSDATAFORMAT_SUBTYPE_IEEE_FLOAT = windows.GUID{0x00000003, 0x0000, 0x0010, [...]byte{0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71}}
+	_KSDATAFORMAT_SUBTYPE_PCM        = windows.GUID{0x00000001, 0x0000, 0x0010, [...]byte{0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71}}
+)
+
+type _AUDCLNT_ERR uint32
+
+const (
+	_AUDCLNT_E_DEVICE_INVALIDATED    _AUDCLNT_ERR = 0x88890004
+	_AUDCLNT_E_NOT_INITIALIZED       _AUDCLNT_ERR = 0x88890001
+	_AUDCLNT_E_RESOURCES_INVALIDATED _AUDCLNT_ERR = 0x88890026
+)
+
+func isAudclntErr(hresult uint32) bool {
+	return hresult&0xffff0000 == (1<<31)|(windows.FACILITY_AUDCLNT<<16)
+}
+
+func (e _AUDCLNT_ERR) Error() string {
+	switch e {
+	case _AUDCLNT_E_DEVICE_INVALIDATED:
+		return "AUDCLNT_E_DEVICE_INVALIDATED"
+	case _AUDCLNT_E_RESOURCES_INVALIDATED:
+		return "AUDCLNT_E_RESOURCES_INVALIDATED"
+	default:
+		return fmt.Sprintf("AUDCLNT_ERR(%d)", e)
+	}
+}
+
+type _AUDCLNT_SHAREMODE int32
+
+const (
+	_AUDCLNT_SHAREMODE_SHARED    _AUDCLNT_SHAREMODE = 0
+	_AUDCLNT_SHAREMODE_EXCLUSIVE _AUDCLNT_SHAREMODE = 1
+)
+
+type _AUDCLNT_STREAMOPTIONS int32
+
+const (
+	_AUDCLNT_STREAMOPTIONS_NONE         _AUDCLNT_STREAMOPTIONS = 0x0
+	_AUDCLNT_STREAMOPTIONS_RAW          _AUDCLNT_STREAMOPTIONS = 0x1
+	_AUDCLNT_STREAMOPTIONS_MATCH_FORMAT _AUDCLNT_STREAMOPTIONS = 0x2
+	_AUDCLNT_STREAMOPTIONS_AMBISONICS   _AUDCLNT_STREAMOPTIONS = 0x4
+)
+
+type _AUDIO_STREAM_CATEGORY int32
+
+const (
+	_AudioCategory_Other                  _AUDIO_STREAM_CATEGORY = 0
+	_AudioCategory_ForegroundOnlyMedia    _AUDIO_STREAM_CATEGORY = 1
+	_AudioCategory_BackgroundCapableMedia _AUDIO_STREAM_CATEGORY = 2
+	_AudioCategory_Communications         _AUDIO_STREAM_CATEGORY = 3
+	_AudioCategory_Alerts                 _AUDIO_STREAM_CATEGORY = 4
+	_AudioCategory_SoundEffects           _AUDIO_STREAM_CATEGORY = 5
+	_AudioCategory_GameEffects            _AUDIO_STREAM_CATEGORY = 6
+	_AudioCategory_GameMedia              _AUDIO_STREAM_CATEGORY = 7
+	_AudioCategory_GameChat               _AUDIO_STREAM_CATEGORY = 8
+	_AudioCategory_Speech                 _AUDIO_STREAM_CATEGORY = 9
+	_AudioCategory_Movie                  _AUDIO_STREAM_CATEGORY = 10
+	_AudioCategory_Media                  _AUDIO_STREAM_CATEGORY = 11
+)
+
+type _CLSCTX int32
+
+const (
+	_CLSCTX_INPROC_SERVER  _CLSCTX = 0x00000001
+	_CLSCTX_INPROC_HANDLER _CLSCTX = 0x00000002
+	_CLSCTX_LOCAL_SERVER   _CLSCTX = 0x00000004
+	_CLSCTX_REMOTE_SERVER  _CLSCTX = 0x00000010
+	_CLSCTX_ALL                    = _CLSCTX_INPROC_SERVER | _CLSCTX_INPROC_HANDLER | _CLSCTX_LOCAL_SERVER | _CLSCTX_REMOTE_SERVER
+)
+
+type _EDataFlow int32
+
+const (
+	eRender _EDataFlow = 0
+)
+
+type _ERole int32
+
+const (
+	eConsole _ERole = 0
+)
+
+type _WIN32_ERR uint32
+
+const (
+	_E_NOTFOUND _WIN32_ERR = 0x80070490
+)
+
+func isWin32Err(hresult uint32) bool {
+	return hresult&0xffff0000 == (1<<31)|(windows.FACILITY_WIN32<<16)
+}
+
+func (e _WIN32_ERR) Error() string {
+	switch e {
+	case _E_NOTFOUND:
+		return "E_NOTFOUND"
+	default:
+		return fmt.Sprintf("HRESULT(%d)", e)
+	}
+}
+
+type _AudioClientProperties struct {
+	cbSize     uint32
+	bIsOffload int32
+	eCategory  _AUDIO_STREAM_CATEGORY
+	Options    _AUDCLNT_STREAMOPTIONS
+}
+
+type _PROPVARIANT struct {
+	// TODO: Implmeent this
+}
+
+type _WAVEFORMATEXTENSIBLE struct {
+	wFormatTag      uint16
+	nChannels       uint16
+	nSamplesPerSec  uint32
+	nAvgBytesPerSec uint32
+	nBlockAlign     uint16
+	wBitsPerSample  uint16
+	cbSize          uint16
+	Samples         uint16 // union
+	dwChannelMask   uint32
+	SubFormat       windows.GUID
+}
+
+func _CoCreateInstance(rclsid *windows.GUID, pUnkOuter unsafe.Pointer, dwClsContext uint32, riid *windows.GUID) (unsafe.Pointer, error) {
+	var v unsafe.Pointer
+	r, _, _ := procCoCreateInstance.Call(uintptr(unsafe.Pointer(rclsid)), uintptr(pUnkOuter), uintptr(dwClsContext), uintptr(unsafe.Pointer(riid)), uintptr(unsafe.Pointer(&v)))
+	runtime.KeepAlive(rclsid)
+	runtime.KeepAlive(riid)
+	if uint32(r) != uint32(windows.S_OK) {
+		return nil, fmt.Errorf("oto: CoCreateInstance failed: HRESULT(%d)", uint32(r))
+	}
+	return v, nil
+}
+
+type _IAudioClient2 struct {
+	vtbl *_IAudioClient2_Vtbl
+}
+
+type _IAudioClient2_Vtbl struct {
+	QueryInterface uintptr
+	AddRef         uintptr
+	Release        uintptr
+
+	Initialize          uintptr
+	GetBufferSize       uintptr
+	GetStreamLatency    uintptr
+	GetCurrentPadding   uintptr
+	IsFormatSupported   uintptr
+	GetMixFormat        uintptr
+	GetDevicePeriod     uintptr
+	Start               uintptr
+	Stop                uintptr
+	Reset               uintptr
+	SetEventHandle      uintptr
+	GetService          uintptr
+	IsOffloadCapable    uintptr
+	SetClientProperties uintptr
+	GetBufferSizeLimits uintptr
+}
+
+func (i *_IAudioClient2) GetBufferSize() (uint32, error) {
+	var numBufferFrames uint32
+	r, _, _ := syscall.Syscall(i.vtbl.GetBufferSize, 2, uintptr(unsafe.Pointer(i)), uintptr(unsafe.Pointer(&numBufferFrames)), 0)
+	if uint32(r) != uint32(windows.S_OK) {
+		if isAudclntErr(uint32(r)) {
+			return 0, fmt.Errorf("oto: IAudioClient2::GetBufferSize failed: %w", _AUDCLNT_ERR(r))
+		}
+		return 0, fmt.Errorf("oto: IAudioClient2::GetBufferSize failed: HRESULT(%d)", uint32(r))
+	}
+	return numBufferFrames, nil
+}
+
+func (i *_IAudioClient2) GetCurrentPadding() (uint32, error) {
+	var numPaddingFrames uint32
+	r, _, _ := syscall.Syscall(i.vtbl.GetCurrentPadding, 2, uintptr(unsafe.Pointer(i)), uintptr(unsafe.Pointer(&numPaddingFrames)), 0)
+	if uint32(r) != uint32(windows.S_OK) {
+		if isAudclntErr(uint32(r)) {
+			return 0, fmt.Errorf("oto: IAudioClient2::GetCurrentPadding failed: %w", _AUDCLNT_ERR(r))
+		}
+		return 0, fmt.Errorf("oto: IAudioClient2::GetCurrentPadding failed: HRESULT(%d)", uint32(r))
+	}
+	return numPaddingFrames, nil
+}
+
+func (i *_IAudioClient2) GetDevicePeriod() (_REFERENCE_TIME, _REFERENCE_TIME, error) {
+	var defaultDevicePeriod _REFERENCE_TIME
+	var minimumDevicePeriod _REFERENCE_TIME
+	r, _, _ := syscall.Syscall(i.vtbl.GetDevicePeriod, 3, uintptr(unsafe.Pointer(i)),
+		uintptr(unsafe.Pointer(&defaultDevicePeriod)), uintptr(unsafe.Pointer(&minimumDevicePeriod)))
+	if uint32(r) != uint32(windows.S_OK) {
+		if isAudclntErr(uint32(r)) {
+			return 0, 0, fmt.Errorf("oto: IAudioClient2::GetDevicePeriod failed: %w", _AUDCLNT_ERR(r))
+		}
+		return 0, 0, fmt.Errorf("oto: IAudioClient2::GetDevicePeriod failed: HRESULT(%d)", uint32(r))
+	}
+	return defaultDevicePeriod, minimumDevicePeriod, nil
+}
+
+func (i *_IAudioClient2) GetService(riid *windows.GUID) (unsafe.Pointer, error) {
+	var v unsafe.Pointer
+	r, _, _ := syscall.Syscall(i.vtbl.GetService, 3, uintptr(unsafe.Pointer(i)), uintptr(unsafe.Pointer(riid)), uintptr(unsafe.Pointer(&v)))
+	if uint32(r) != uint32(windows.S_OK) {
+		if isAudclntErr(uint32(r)) {
+			return nil, fmt.Errorf("oto: IAudioClient2::GetService failed: %w", _AUDCLNT_ERR(r))
+		}
+		return nil, fmt.Errorf("oto: IAudioClient2::GetService failed: HRESULT(%d)", uint32(r))
+	}
+	return v, nil
+}
+
+func (i *_IAudioClient2) Initialize(shareMode _AUDCLNT_SHAREMODE, streamFlags uint32, hnsBufferDuration _REFERENCE_TIME, hnsPeriodicity _REFERENCE_TIME, pFormat *_WAVEFORMATEXTENSIBLE, audioSessionGuid *windows.GUID) error {
+	var r uintptr
+	if unsafe.Sizeof(uintptr(0)) == 8 {
+		// 64bits
+		r, _, _ = syscall.Syscall9(i.vtbl.Initialize, 7, uintptr(unsafe.Pointer(i)),
+			uintptr(shareMode), uintptr(streamFlags), uintptr(hnsBufferDuration),
+			uintptr(hnsPeriodicity), uintptr(unsafe.Pointer(pFormat)), uintptr(unsafe.Pointer(audioSessionGuid)),
+			0, 0)
+	} else {
+		// 32bits
+		r, _, _ = syscall.Syscall9(i.vtbl.Initialize, 9, uintptr(unsafe.Pointer(i)),
+			uintptr(shareMode), uintptr(streamFlags), uintptr(hnsBufferDuration),
+			uintptr(hnsBufferDuration>>32), uintptr(hnsPeriodicity), uintptr(hnsPeriodicity>>32),
+			uintptr(unsafe.Pointer(pFormat)), uintptr(unsafe.Pointer(audioSessionGuid)))
+	}
+	runtime.KeepAlive(pFormat)
+	runtime.KeepAlive(audioSessionGuid)
+	if uint32(r) != uint32(windows.S_OK) {
+		if isAudclntErr(uint32(r)) {
+			return fmt.Errorf("oto: IAudioClient2::Initialize failed: %w", _AUDCLNT_ERR(r))
+		}
+		return fmt.Errorf("oto: IAudioClient2::Initialize failed: HRESULT(%d)", uint32(r))
+	}
+	return nil
+}
+
+func (i *_IAudioClient2) IsFormatSupported(shareMode _AUDCLNT_SHAREMODE, pFormat *_WAVEFORMATEXTENSIBLE) (*_WAVEFORMATEXTENSIBLE, error) {
+	var closestMatch *_WAVEFORMATEXTENSIBLE
+	r, _, _ := syscall.Syscall6(i.vtbl.IsFormatSupported, 4, uintptr(unsafe.Pointer(i)),
+		uintptr(shareMode), uintptr(unsafe.Pointer(pFormat)), uintptr(unsafe.Pointer(&closestMatch)),
+		0, 0)
+	if uint32(r) != uint32(windows.S_OK) {
+		if uint32(r) == uint32(windows.S_FALSE) {
+			var r _WAVEFORMATEXTENSIBLE
+			if closestMatch != nil {
+				r = *closestMatch
+				windows.CoTaskMemFree(unsafe.Pointer(closestMatch))
+			}
+			return &r, nil
+		}
+		if isAudclntErr(uint32(r)) {
+			return nil, fmt.Errorf("oto: IAudioClient2::IsFormatSupported failed: %w", _AUDCLNT_ERR(r))
+		}
+		return nil, fmt.Errorf("oto: IAudioClient2::IsFormatSupported failed: HRESULT(%d)", uint32(r))
+	}
+	return nil, nil
+}
+
+func (i *_IAudioClient2) Release() uint32 {
+	r, _, _ := syscall.Syscall(i.vtbl.Release, 1, uintptr(unsafe.Pointer(i)), 0, 0)
+	return uint32(r)
+}
+
+func (i *_IAudioClient2) SetClientProperties(pProperties *_AudioClientProperties) error {
+	r, _, _ := syscall.Syscall(i.vtbl.SetClientProperties, 2, uintptr(unsafe.Pointer(i)), uintptr(unsafe.Pointer(pProperties)), 0)
+	runtime.KeepAlive(pProperties)
+	if uint32(r) != uint32(windows.S_OK) {
+		if isAudclntErr(uint32(r)) {
+			return fmt.Errorf("oto: IAudioClient2::SetClientProperties failed: %w", _AUDCLNT_ERR(r))
+		}
+		return fmt.Errorf("oto: IAudioClient2::SetClientProperties failed: HRESULT(%d)", uint32(r))
+	}
+	return nil
+}
+
+func (i *_IAudioClient2) SetEventHandle(eventHandle windows.Handle) error {
+	r, _, _ := syscall.Syscall(i.vtbl.SetEventHandle, 2, uintptr(unsafe.Pointer(i)), uintptr(eventHandle), 0)
+	if uint32(r) != uint32(windows.S_OK) {
+		if isAudclntErr(uint32(r)) {
+			return fmt.Errorf("oto: IAudioClient2::SetEventHandle failed: %w", _AUDCLNT_ERR(r))
+		}
+		return fmt.Errorf("oto: IAudioClient2::SetEventHandle failed: HRESULT(%d)", uint32(r))
+	}
+	return nil
+}
+
+func (i *_IAudioClient2) Start() error {
+	r, _, _ := syscall.Syscall(i.vtbl.Start, 1, uintptr(unsafe.Pointer(i)), 0, 0)
+	if uint32(r) != uint32(windows.S_OK) {
+		if isAudclntErr(uint32(r)) {
+			return fmt.Errorf("oto: IAudioClient2::Start failed: %w", _AUDCLNT_ERR(r))
+		}
+		return fmt.Errorf("oto: IAudioClient2::Start failed: HRESULT(%d)", uint32(r))
+	}
+	return nil
+}
+
+func (i *_IAudioClient2) Stop() (bool, error) {
+	r, _, _ := syscall.Syscall(i.vtbl.Stop, 1, uintptr(unsafe.Pointer(i)), 0, 0)
+	if uint32(r) != uint32(windows.S_OK) && uint32(r) != uint32(windows.S_FALSE) {
+		if isAudclntErr(uint32(r)) {
+			return false, fmt.Errorf("oto: IAudioClient2::Stop failed: %w", _AUDCLNT_ERR(r))
+		}
+		return false, fmt.Errorf("oto: IAudioClient2::Stop failed: HRESULT(%d)", uint32(r))
+	}
+	return uint32(r) == uint32(windows.S_OK), nil
+}
+
+type _IAudioRenderClient struct {
+	vtbl *_IAudioRenderClient_Vtbl
+}
+
+type _IAudioRenderClient_Vtbl struct {
+	QueryInterface uintptr
+	AddRef         uintptr
+	Release        uintptr
+
+	GetBuffer     uintptr
+	ReleaseBuffer uintptr
+}
+
+func (i *_IAudioRenderClient) GetBuffer(numFramesRequested uint32) (*byte, error) {
+	var data *byte
+	r, _, _ := syscall.Syscall(i.vtbl.GetBuffer, 3, uintptr(unsafe.Pointer(i)), uintptr(numFramesRequested), uintptr(unsafe.Pointer(&data)))
+	if uint32(r) != uint32(windows.S_OK) {
+		if isAudclntErr(uint32(r)) {
+			return nil, fmt.Errorf("oto: IAudioRenderClient::GetBuffer failed: %w", _AUDCLNT_ERR(r))
+		}
+		return nil, fmt.Errorf("oto: IAudioRenderClient::GetBuffer failed: HRESULT(%d)", uint32(r))
+	}
+	return data, nil
+}
+
+func (i *_IAudioRenderClient) Release() uint32 {
+	r, _, _ := syscall.Syscall(i.vtbl.Release, 1, uintptr(unsafe.Pointer(i)), 0, 0)
+	return uint32(r)
+}
+
+func (i *_IAudioRenderClient) ReleaseBuffer(numFramesWritten uint32, dwFlags uint32) error {
+	r, _, _ := syscall.Syscall(i.vtbl.ReleaseBuffer, 3, uintptr(unsafe.Pointer(i)), uintptr(numFramesWritten), uintptr(dwFlags))
+	if uint32(r) != uint32(windows.S_OK) {
+		if isAudclntErr(uint32(r)) {
+			return fmt.Errorf("oto: IAudioRenderClient::ReleaseBuffer failed: %w", _AUDCLNT_ERR(r))
+		}
+		return fmt.Errorf("oto: IAudioRenderClient::ReleaseBuffer failed: HRESULT(%d)", uint32(r))
+	}
+	return nil
+}
+
+type _IMMDevice struct {
+	vtbl *_IMMDevice_Vtbl
+}
+
+type _IMMDevice_Vtbl struct {
+	QueryInterface uintptr
+	AddRef         uintptr
+	Release        uintptr
+
+	Activate          uintptr
+	OpenPropertyStore uintptr
+	GetId             uintptr
+	GetState          uintptr
+}
+
+func (i *_IMMDevice) Activate(iid *windows.GUID, dwClsCtx uint32, pActivationParams *_PROPVARIANT) (unsafe.Pointer, error) {
+	var v unsafe.Pointer
+	r, _, _ := syscall.Syscall6(i.vtbl.Activate, 5, uintptr(unsafe.Pointer(i)),
+		uintptr(unsafe.Pointer(iid)), uintptr(dwClsCtx), uintptr(unsafe.Pointer(pActivationParams)), uintptr(unsafe.Pointer(&v)), 0)
+	runtime.KeepAlive(iid)
+	runtime.KeepAlive(pActivationParams)
+	if uint32(r) != uint32(windows.S_OK) {
+		return nil, fmt.Errorf("oto: IMMDevice::Activate failed: HRESULT(%d)", uint32(r))
+	}
+	return v, nil
+}
+
+func (i *_IMMDevice) GetId() (string, error) {
+	var strId *uint16
+	r, _, _ := syscall.Syscall(i.vtbl.GetId, 2, uintptr(unsafe.Pointer(i)), uintptr(unsafe.Pointer(&strId)), 0)
+	if uint32(r) != uint32(windows.S_OK) {
+		return "", fmt.Errorf("oto: IMMDevice::GetId failed: HRESULT(%d)", uint32(r))
+	}
+	return windows.UTF16PtrToString(strId), nil
+}
+
+func (i *_IMMDevice) Release() {
+	syscall.Syscall(i.vtbl.Release, 1, uintptr(unsafe.Pointer(i)), 0, 0)
+}
+
+type _IMMDeviceEnumerator struct {
+	vtbl *_IMMDeviceEnumerator_Vtbl
+}
+
+type _IMMDeviceEnumerator_Vtbl struct {
+	QueryInterface uintptr
+	AddRef         uintptr
+	Release        uintptr
+
+	EnumAudioEndpoints                     uintptr
+	GetDefaultAudioEndpoint                uintptr
+	GetDevice                              uintptr
+	RegisterEndpointNotificationCallback   uintptr
+	UnregisterEndpointNotificationCallback uintptr
+}
+
+func (i *_IMMDeviceEnumerator) GetDefaultAudioEndPoint(dataFlow _EDataFlow, role _ERole) (*_IMMDevice, error) {
+	var endPoint *_IMMDevice
+	r, _, _ := syscall.Syscall6(i.vtbl.GetDefaultAudioEndpoint, 4, uintptr(unsafe.Pointer(i)),
+		uintptr(dataFlow), uintptr(role), uintptr(unsafe.Pointer(&endPoint)), 0, 0)
+	if uint32(r) != uint32(windows.S_OK) {
+		if isWin32Err(uint32(r)) {
+			return nil, fmt.Errorf("oto: IMMDeviceEnumerator::GetDefaultAudioEndPoint failed: %w", _E_NOTFOUND)
+		}
+		return nil, fmt.Errorf("oto: IMMDeviceEnumerator::GetDefaultAudioEndPoint failed: HRESULT(%d)", uint32(r))
+	}
+	return endPoint, nil
+}
+
+func (i *_IMMDeviceEnumerator) Release() {
+	syscall.Syscall(i.vtbl.Release, 1, uintptr(unsafe.Pointer(i)), 0, 0)
+}

vendor/github.com/ebitengine/oto/v3/api_winmm_windows.go

@@ -0,0 +1,174 @@
+// Copyright 2021 The Oto Authors
+//
+// Licensed under the Apache License, Version 2.0 (the "License");
+// you may not use this file except in compliance with the License.
+// You may obtain a copy of the License at
+//
+//     http://www.apache.org/licenses/LICENSE-2.0
+//
+// Unless required by applicable law or agreed to in writing, software
+// distributed under the License is distributed on an "AS IS" BASIS,
+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+// See the License for the specific language governing permissions and
+// limitations under the License.
+
+package oto
+
+import (
+	"fmt"
+	"runtime"
+	"unsafe"
+
+	"golang.org/x/sys/windows"
+)
+
+var (
+	winmm = windows.NewLazySystemDLL("winmm")
+)
+
+var (
+	procWaveOutOpen            = winmm.NewProc("waveOutOpen")
+	procWaveOutClose           = winmm.NewProc("waveOutClose")
+	procWaveOutPrepareHeader   = winmm.NewProc("waveOutPrepareHeader")
+	procWaveOutUnprepareHeader = winmm.NewProc("waveOutUnprepareHeader")
+	procWaveOutWrite           = winmm.NewProc("waveOutWrite")
+)
+
+type _WAVEHDR struct {
+	lpData          uintptr
+	dwBufferLength  uint32
+	dwBytesRecorded uint32
+	dwUser          uintptr
+	dwFlags         uint32
+	dwLoops         uint32
+	lpNext          uintptr
+	reserved        uintptr
+}
+
+type _WAVEFORMATEX struct {
+	wFormatTag      uint16
+	nChannels       uint16
+	nSamplesPerSec  uint32
+	nAvgBytesPerSec uint32
+	nBlockAlign     uint16
+	wBitsPerSample  uint16
+	cbSize          uint16
+}
+
+const (
+	_WAVE_FORMAT_IEEE_FLOAT = 3
+	_WHDR_INQUEUE           = 16
+)
+
+type _MMRESULT uint
+
+const (
+	_MMSYSERR_NOERROR       _MMRESULT = 0
+	_MMSYSERR_ERROR         _MMRESULT = 1
+	_MMSYSERR_BADDEVICEID   _MMRESULT = 2
+	_MMSYSERR_ALLOCATED     _MMRESULT = 4
+	_MMSYSERR_INVALIDHANDLE _MMRESULT = 5
+	_MMSYSERR_NODRIVER      _MMRESULT = 6
+	_MMSYSERR_NOMEM         _MMRESULT = 7
+	_WAVERR_BADFORMAT       _MMRESULT = 32
+	_WAVERR_STILLPLAYING    _MMRESULT = 33
+	_WAVERR_UNPREPARED      _MMRESULT = 34
+	_WAVERR_SYNC            _MMRESULT = 35
+)
+
+func (m _MMRESULT) Error() string {
+	switch m {
+	case _MMSYSERR_NOERROR:
+		return "MMSYSERR_NOERROR"
+	case _MMSYSERR_ERROR:
+		return "MMSYSERR_ERROR"
+	case _MMSYSERR_BADDEVICEID:
+		return "MMSYSERR_BADDEVICEID"
+	case _MMSYSERR_ALLOCATED:
+		return "MMSYSERR_ALLOCATED"
+	case _MMSYSERR_INVALIDHANDLE:
+		return "MMSYSERR_INVALIDHANDLE"
+	case _MMSYSERR_NODRIVER:
+		return "MMSYSERR_NODRIVER"
+	case _MMSYSERR_NOMEM:
+		return "MMSYSERR_NOMEM"
+	case _WAVERR_BADFORMAT:
+		return "WAVERR_BADFORMAT"
+	case _WAVERR_STILLPLAYING:
+		return "WAVERR_STILLPLAYING"
+	case _WAVERR_UNPREPARED:
+		return "WAVERR_UNPREPARED"
+	case _WAVERR_SYNC:
+		return "WAVERR_SYNC"
+	}
+	return fmt.Sprintf("MMRESULT (%d)", m)
+}
+
+func waveOutOpen(f *_WAVEFORMATEX, callback uintptr) (uintptr, error) {
+	const (
+		waveMapper       = 0xffffffff
+		callbackFunction = 0x30000
+	)
+	var w uintptr
+	var fdwOpen uintptr
+	if callback != 0 {
+		fdwOpen |= callbackFunction
+	}
+	r, _, e := procWaveOutOpen.Call(uintptr(unsafe.Pointer(&w)), waveMapper, uintptr(unsafe.Pointer(f)),
+		callback, 0, fdwOpen)
+	runtime.KeepAlive(f)
+	if _MMRESULT(r) != _MMSYSERR_NOERROR {
+		if e != nil && e != windows.ERROR_SUCCESS {
+			return 0, fmt.Errorf("oto: waveOutOpen failed: %w", e)
+		}
+		return 0, fmt.Errorf("oto: waveOutOpen failed: %w", _MMRESULT(r))
+	}
+	return w, nil
+}
+
+func waveOutClose(hwo uintptr) error {
+	r, _, e := procWaveOutClose.Call(hwo)
+	if _MMRESULT(r) != _MMSYSERR_NOERROR {
+		if e != nil && e != windows.ERROR_SUCCESS {
+			return fmt.Errorf("oto: waveOutClose failed: %w", e)
+		}
+		return fmt.Errorf("oto: waveOutClose failed: %w", _MMRESULT(r))
+	}
+	return nil
+}
+
+func waveOutPrepareHeader(hwo uintptr, pwh *_WAVEHDR) error {
+	r, _, e := procWaveOutPrepareHeader.Call(hwo, uintptr(unsafe.Pointer(pwh)), unsafe.Sizeof(_WAVEHDR{}))
+	runtime.KeepAlive(pwh)
+	if _MMRESULT(r) != _MMSYSERR_NOERROR {
+		if e != nil && e != windows.ERROR_SUCCESS {
+			return fmt.Errorf("oto: waveOutPrepareHeader failed: %w", e)
+		}
+		return fmt.Errorf("oto: waveOutPrepareHeader failed: %w", _MMRESULT(r))
+	}
+	return nil
+}
+
+func waveOutUnprepareHeader(hwo uintptr, pwh *_WAVEHDR) error {
+	r, _, e := procWaveOutUnprepareHeader.Call(hwo, uintptr(unsafe.Pointer(pwh)), unsafe.Sizeof(_WAVEHDR{}))
+	runtime.KeepAlive(pwh)
+	if _MMRESULT(r) != _MMSYSERR_NOERROR {
+		if e != nil && e != windows.ERROR_SUCCESS {
+			return fmt.Errorf("oto: waveOutUnprepareHeader failed: %w", e)
+		}
+		return fmt.Errorf("oto: waveOutUnprepareHeader failed: %w", _MMRESULT(r))
+	}
+	return nil
+}
+
+func waveOutWrite(hwo uintptr, pwh *_WAVEHDR) error {
+	r, _, e := procWaveOutWrite.Call(hwo, uintptr(unsafe.Pointer(pwh)), unsafe.Sizeof(_WAVEHDR{}))
+	runtime.KeepAlive(pwh)
+	if _MMRESULT(r) != _MMSYSERR_NOERROR {
+		if e != nil && e != windows.ERROR_SUCCESS {
+			return fmt.Errorf("oto: waveOutWrite failed: %w", e)
+		}
+		return fmt.Errorf("oto: waveOutWrite failed: %w", _MMRESULT(r))
+	}
+	return nil
+}

vendor/github.com/ebitengine/oto/v3/context.go

@@ -0,0 +1,181 @@
+// Copyright 2021 The Oto Authors
+//
+// Licensed under the Apache License, Version 2.0 (the "License");
+// you may not use this file except in compliance with the License.
+// You may obtain a copy of the License at
+//
+//     http://www.apache.org/licenses/LICENSE-2.0
+//
+// Unless required by applicable law or agreed to in writing, software
+// distributed under the License is distributed on an "AS IS" BASIS,
+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+// See the License for the specific language governing permissions and
+// limitations under the License.
+
+package oto
+
+import (
+	"fmt"
+	"io"
+	"sync"
+	"time"
+
+	"github.com/ebitengine/oto/v3/internal/mux"
+)
+
+var (
+	contextCreated       bool
+	contextCreationMutex sync.Mutex
+)
+
+// Context is the main object in Oto. It interacts with the audio drivers.
+//
+// To play sound with Oto, first create a context. Then use the context to create
+// an arbitrary number of players. Then use the players to play sound.
+//
+// Creating multiple contexts is NOT supported.
+type Context struct {
+	context *context
+}
+
+// Format is the format of sources.
+type Format int
+
+const (
+	// FormatFloat32LE is the format of 32 bits floats little endian.
+	FormatFloat32LE Format = iota
+
+	// FormatUnsignedInt8 is the format of 8 bits integers.
+	FormatUnsignedInt8
+
+	//FormatSignedInt16LE is the format of 16 bits integers little endian.
+	FormatSignedInt16LE
+)
+
+// NewContextOptions represents options for NewContext.
+type NewContextOptions struct {
+	// SampleRate specifies the number of samples that should be played during one second.
+	// Usual numbers are 44100 or 48000. One context has only one sample rate. You cannot play multiple audio
+	// sources with different sample rates at the same time.
+	SampleRate int
+
+	// ChannelCount specifies the number of channels. One channel is mono playback. Two
+	// channels are stereo playback. No other values are supported.
+	ChannelCount int
+
+	// Format specifies the format of sources.
+	Format Format
+
+	// BufferSize specifies a buffer size in the underlying device.
+	//
+	// If 0 is specified, the driver's default buffer size is used.
+	// Set BufferSize to adjust the buffer size if you want to adjust latency or reduce noises.
+	// Too big buffer size can increase the latency time.
+	// On the other hand, too small buffer size can cause glitch noises due to buffer shortage.
+	BufferSize time.Duration
+}
+
+// NewContext creates a new context with given options.
+// A context creates and holds ready-to-use Player objects.
+// NewContext returns a context, a channel that is closed when the context is ready, and an error if it exists.
+//
+// Creating multiple contexts is NOT supported.
+func NewContext(options *NewContextOptions) (*Context, chan struct{}, error) {
+	contextCreationMutex.Lock()
+	defer contextCreationMutex.Unlock()
+
+	if contextCreated {
+		return nil, nil, fmt.Errorf("oto: context is already created")
+	}
+	contextCreated = true
+
+	var bufferSizeInBytes int
+	if options.BufferSize != 0 {
+		// The underying driver always uses 32bit floats.
+		bytesPerSample := options.ChannelCount * 4
+		bytesPerSecond := options.SampleRate * bytesPerSample
+		bufferSizeInBytes = int(int64(options.BufferSize) * int64(bytesPerSecond) / int64(time.Second))
+		bufferSizeInBytes = bufferSizeInBytes / bytesPerSample * bytesPerSample
+	}
+	ctx, ready, err := newContext(options.SampleRate, options.ChannelCount, mux.Format(options.Format), bufferSizeInBytes)
+	if err != nil {
+		return nil, nil, err
+	}
+	return &Context{context: ctx}, ready, nil
+}
+
+// NewPlayer creates a new, ready-to-use Player belonging to the Context.
+// It is safe to create multiple players.
+//
+// The format of r is as follows:
+//
+//	[data]      = [sample 1] [sample 2] [sample 3] ...
+//	[sample *]  = [channel 1] [channel 2] ...
+//	[channel *] = [byte 1] [byte 2] ...
+//
+// Byte ordering is little endian.
+//
+// A player has some amount of an underlying buffer.
+// Read data from r is queued to the player's underlying buffer.
+// The underlying buffer is consumed by its playing.
+// Then, r's position and the current playing position don't necessarily match.
+// If you want to clear the underlying buffer for some reasons e.g., you want to seek the position of r,
+// call the player's Reset function.
+//
+// You cannot share r by multiple players.
+//
+// The returned player implements Player, BufferSizeSetter, and io.Seeker.
+// You can modify the buffer size of a player by the SetBufferSize function.
+// A small buffer size is useful if you want to play a real-time PCM for example.
+// Note that the audio quality might be affected if you modify the buffer size.
+//
+// If r does not implement io.Seeker, the returned player's Seek returns an error.
+//
+// NewPlayer is concurrent-safe.
+//
+// All the functions of a Player returned by NewPlayer are concurrent-safe.
+func (c *Context) NewPlayer(r io.Reader) *Player {
+	return &Player{
+		player: c.context.mux.NewPlayer(r),
+	}
+}
+
+// Suspend suspends the entire audio play.
+//
+// Suspend is concurrent-safe.
+func (c *Context) Suspend() error {
+	return c.context.Suspend()
+}
+
+// Resume resumes the entire audio play, which was suspended by Suspend.
+//
+// Resume is concurrent-safe.
+func (c *Context) Resume() error {
+	return c.context.Resume()
+}
+
+// Err returns the current error.
+//
+// Err is concurrent-safe.
+func (c *Context) Err() error {
+	return c.context.Err()
+}
+
+type atomicError struct {
+	err error
+	m   sync.Mutex
+}
+
+func (a *atomicError) TryStore(err error) {
+	a.m.Lock()
+	defer a.m.Unlock()
+	if a.err == nil {
+		a.err = err
+	}
+}
+
+func (a *atomicError) Load() error {
+	a.m.Lock()
+	defer a.m.Unlock()
+	return a.err
+}

vendor/github.com/ebitengine/oto/v3/driver_android.go

@@ -0,0 +1,49 @@
+// Copyright 2021 The Oto Authors
+//
+// Licensed under the Apache License, Version 2.0 (the "License");
+// you may not use this file except in compliance with the License.
+// You may obtain a copy of the License at
+//
+//     http://www.apache.org/licenses/LICENSE-2.0
+//
+// Unless required by applicable law or agreed to in writing, software
+// distributed under the License is distributed on an "AS IS" BASIS,
+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+// See the License for the specific language governing permissions and
+// limitations under the License.
+
+package oto
+
+import (
+	"github.com/ebitengine/oto/v3/internal/mux"
+	"github.com/ebitengine/oto/v3/internal/oboe"
+)
+
+type context struct {
+	mux *mux.Mux
+}
+
+func newContext(sampleRate int, channelCount int, format mux.Format, bufferSizeInBytes int) (*context, chan struct{}, error) {
+	ready := make(chan struct{})
+	close(ready)
+
+	c := &context{
+		mux: mux.New(sampleRate, channelCount, format),
+	}
+	if err := oboe.Play(sampleRate, channelCount, c.mux.ReadFloat32s, bufferSizeInBytes); err != nil {
+		return nil, nil, err
+	}
+	return c, ready, nil
+}
+
+func (c *context) Suspend() error {
+	return oboe.Suspend()
+}
+
+func (c *context) Resume() error {
+	return oboe.Resume()
+}
+
+func (c *context) Err() error {
+	return nil
+}

vendor/github.com/ebitengine/oto/v3/driver_darwin.go

@@ -0,0 +1,263 @@
+// Copyright 2021 The Oto Authors
+//
+// Licensed under the Apache License, Version 2.0 (the "License");
+// you may not use this file except in compliance with the License.
+// You may obtain a copy of the License at
+//
+//     http://www.apache.org/licenses/LICENSE-2.0
+//
+// Unless required by applicable law or agreed to in writing, software
+// distributed under the License is distributed on an "AS IS" BASIS,
+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+// See the License for the specific language governing permissions and
+// limitations under the License.
+
+package oto
+
+import (
+	"fmt"
+	"sync"
+	"time"
+	"unsafe"
+
+	"github.com/ebitengine/purego/objc"
+
+	"github.com/ebitengine/oto/v3/internal/mux"
+)
+
+const (
+	float32SizeInBytes = 4
+
+	bufferCount = 4
+
+	noErr = 0
+)
+
+func newAudioQueue(sampleRate, channelCount int, oneBufferSizeInBytes int) (_AudioQueueRef, []_AudioQueueBufferRef, error) {
+	desc := _AudioStreamBasicDescription{
+		mSampleRate:       float64(sampleRate),
+		mFormatID:         uint32(kAudioFormatLinearPCM),
+		mFormatFlags:      uint32(kAudioFormatFlagIsFloat),
+		mBytesPerPacket:   uint32(channelCount * float32SizeInBytes),
+		mFramesPerPacket:  1,
+		mBytesPerFrame:    uint32(channelCount * float32SizeInBytes),
+		mChannelsPerFrame: uint32(channelCount),
+		mBitsPerChannel:   uint32(8 * float32SizeInBytes),
+	}
+
+	var audioQueue _AudioQueueRef
+	if osstatus := _AudioQueueNewOutput(
+		&desc,
+		render,
+		nil,
+		0, //CFRunLoopRef
+		0, //CFStringRef
+		0,
+		&audioQueue); osstatus != noErr {
+		return 0, nil, fmt.Errorf("oto: AudioQueueNewFormat with StreamFormat failed: %d", osstatus)
+	}
+
+	bufs := make([]_AudioQueueBufferRef, 0, bufferCount)
+	for len(bufs) < cap(bufs) {
+		var buf _AudioQueueBufferRef
+		if osstatus := _AudioQueueAllocateBuffer(audioQueue, uint32(oneBufferSizeInBytes), &buf); osstatus != noErr {
+			return 0, nil, fmt.Errorf("oto: AudioQueueAllocateBuffer failed: %d", osstatus)
+		}
+		buf.mAudioDataByteSize = uint32(oneBufferSizeInBytes)
+		bufs = append(bufs, buf)
+	}
+
+	return audioQueue, bufs, nil
+}
+
+type context struct {
+	audioQueue      _AudioQueueRef
+	unqueuedBuffers []_AudioQueueBufferRef
+
+	oneBufferSizeInBytes int
+
+	cond *sync.Cond
+
+	mux *mux.Mux
+	err atomicError
+}
+
+// TODO: Convert the error code correctly.
+// See https://stackoverflow.com/questions/2196869/how-do-you-convert-an-iphone-osstatus-code-to-something-useful
+
+var theContext *context
+
+func newContext(sampleRate int, channelCount int, format mux.Format, bufferSizeInBytes int) (*context, chan struct{}, error) {
+	var oneBufferSizeInBytes int
+	if bufferSizeInBytes != 0 {
+		oneBufferSizeInBytes = bufferSizeInBytes / bufferCount
+	} else {
+		oneBufferSizeInBytes = defaultOneBufferSizeInBytes
+	}
+	bytesPerSample := channelCount * 4
+	oneBufferSizeInBytes = oneBufferSizeInBytes / bytesPerSample * bytesPerSample
+
+	ready := make(chan struct{})
+
+	c := &context{
+		cond:                 sync.NewCond(&sync.Mutex{}),
+		mux:                  mux.New(sampleRate, channelCount, format),
+		oneBufferSizeInBytes: oneBufferSizeInBytes,
+	}
+	theContext = c
+
+	if err := initializeAPI(); err != nil {
+		return nil, nil, err
+	}
+
+	go func() {
+		defer close(ready)
+
+		q, bs, err := newAudioQueue(sampleRate, channelCount, oneBufferSizeInBytes)
+		if err != nil {
+			c.err.TryStore(err)
+			return
+		}
+		c.audioQueue = q
+		c.unqueuedBuffers = bs
+
+		if err := setNotificationHandler(); err != nil {
+			c.err.TryStore(err)
+			return
+		}
+
+		var retryCount int
+	try:
+		if osstatus := _AudioQueueStart(c.audioQueue, nil); osstatus != noErr {
+			if osstatus == avAudioSessionErrorCodeCannotStartPlaying && retryCount < 100 {
+				// TODO: use sleepTime() after investigating when this error happens.
+				time.Sleep(10 * time.Millisecond)
+				retryCount++
+				goto try
+			}
+			c.err.TryStore(fmt.Errorf("oto: AudioQueueStart failed at newContext: %d", osstatus))
+			return
+		}
+
+		go c.loop()
+	}()
+
+	return c, ready, nil
+}
+
+func (c *context) wait() bool {
+	c.cond.L.Lock()
+	defer c.cond.L.Unlock()
+
+	for len(c.unqueuedBuffers) == 0 && c.err.Load() == nil {
+		c.cond.Wait()
+	}
+	return c.err.Load() == nil
+}
+
+func (c *context) loop() {
+	buf32 := make([]float32, c.oneBufferSizeInBytes/4)
+	for {
+		if !c.wait() {
+			return
+		}
+		c.appendBuffer(buf32)
+	}
+}
+
+func (c *context) appendBuffer(buf32 []float32) {
+	c.cond.L.Lock()
+	defer c.cond.L.Unlock()
+
+	if c.err.Load() != nil {
+		return
+	}
+
+	buf := c.unqueuedBuffers[0]
+	copy(c.unqueuedBuffers, c.unqueuedBuffers[1:])
+	c.unqueuedBuffers = c.unqueuedBuffers[:len(c.unqueuedBuffers)-1]
+
+	c.mux.ReadFloat32s(buf32)
+	copy(unsafe.Slice((*float32)(unsafe.Pointer(buf.mAudioData)), buf.mAudioDataByteSize/float32SizeInBytes), buf32)
+
+	if osstatus := _AudioQueueEnqueueBuffer(c.audioQueue, buf, 0, nil); osstatus != noErr {
+		c.err.TryStore(fmt.Errorf("oto: AudioQueueEnqueueBuffer failed: %d", osstatus))
+	}
+}
+
+func (c *context) Suspend() error {
+	c.cond.L.Lock()
+	defer c.cond.L.Unlock()
+
+	if err := c.err.Load(); err != nil {
+		return err.(error)
+	}
+	if osstatus := _AudioQueuePause(c.audioQueue); osstatus != noErr {
+		return fmt.Errorf("oto: AudioQueuePause failed: %d", osstatus)
+	}
+	return nil
+}
+
+func (c *context) Resume() error {
+	c.cond.L.Lock()
+	defer c.cond.L.Unlock()
+
+	if err := c.err.Load(); err != nil {
+		return err.(error)
+	}
+
+	var retryCount int
+try:
+	if osstatus := _AudioQueueStart(c.audioQueue, nil); osstatus != noErr {
+		if (osstatus == avAudioSessionErrorCodeCannotStartPlaying ||
+			osstatus == avAudioSessionErrorCodeCannotInterruptOthers) &&
+			retryCount < 30 {
+			// It is uncertain that this error is temporary or not. Then let's use exponential-time sleeping.
+			time.Sleep(sleepTime(retryCount))
+			retryCount++
+			goto try
+		}
+		if osstatus == avAudioSessionErrorCodeSiriIsRecording {
+			// As this error should be temporary, it should be OK to use a short time for sleep anytime.
+			time.Sleep(10 * time.Millisecond)
+			goto try
+		}
+		return fmt.Errorf("oto: AudioQueueStart failed at Resume: %d", osstatus)
+	}
+	return nil
+}
+
+func (c *context) Err() error {
+	if err := c.err.Load(); err != nil {
+		return err.(error)
+	}
+	return nil
+}
+
+func render(inUserData unsafe.Pointer, inAQ _AudioQueueRef, inBuffer _AudioQueueBufferRef) {
+	theContext.cond.L.Lock()
+	defer theContext.cond.L.Unlock()
+	theContext.unqueuedBuffers = append(theContext.unqueuedBuffers, inBuffer)
+	theContext.cond.Signal()
+}
+
+func setGlobalPause(self objc.ID, _cmd objc.SEL, notification objc.ID) {
+	theContext.Suspend()
+}
+
+func setGlobalResume(self objc.ID, _cmd objc.SEL, notification objc.ID) {
+	theContext.Resume()
+}
+
+func sleepTime(count int) time.Duration {
+	switch count {
+	case 0:
+		return 10 * time.Millisecond
+	case 1:
+		return 20 * time.Millisecond
+	case 2:
+		return 50 * time.Millisecond
+	default:
+		return 100 * time.Millisecond
+	}
+}

vendor/github.com/ebitengine/oto/v3/driver_ios.go

@@ -0,0 +1,30 @@
+// Copyright 2021 The Oto Authors
+//
+// Licensed under the Apache License, Version 2.0 (the "License");
+// you may not use this file except in compliance with the License.
+// You may obtain a copy of the License at
+//
+//     http://www.apache.org/licenses/LICENSE-2.0
+//
+// Unless required by applicable law or agreed to in writing, software
+// distributed under the License is distributed on an "AS IS" BASIS,
+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+// See the License for the specific language governing permissions and
+// limitations under the License.
+
+//go:build darwin && ios
+
+package oto
+
+// 12288 seems necessary at least on iPod touch (7th).
+// With 48000[Hz] stereo, the maximum delay is (12288*4[buffers] / 4 / 2)[samples] / 48000 [Hz] = 100[ms].
+// '4' is float32 size in bytes. '2' is a number of channels for stereo.
+
+const defaultOneBufferSizeInBytes = 12288
+
+func setNotificationHandler() error {
+	// AVAudioSessionInterruptionNotification is not reliable on iOS. Rely on
+	// applicationWillResignActive and applicationDidBecomeActive instead. See
+	// https://stackoverflow.com/questions/24404463/ios-siri-not-available-does-not-return-avaudiosessioninterruptionoptionshouldre
+	return nil
+}

vendor/github.com/ebitengine/oto/v3/driver_js.go

@@ -0,0 +1,141 @@
+// Copyright 2021 The Oto Authors
+//
+// Licensed under the Apache License, Version 2.0 (the "License");
+// you may not use this file except in compliance with the License.
+// You may obtain a copy of the License at
+//
+//     http://www.apache.org/licenses/LICENSE-2.0
+//
+// Unless required by applicable law or agreed to in writing, software
+// distributed under the License is distributed on an "AS IS" BASIS,
+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+// See the License for the specific language governing permissions and
+// limitations under the License.
+
+package oto
+
+import (
+	"errors"
+	"runtime"
+	"syscall/js"
+	"unsafe"
+
+	"github.com/ebitengine/oto/v3/internal/mux"
+)
+
+type context struct {
+	audioContext            js.Value
+	scriptProcessor         js.Value
+	scriptProcessorCallback js.Func
+	ready                   bool
+	callbacks               map[string]js.Func
+
+	mux *mux.Mux
+}
+
+func newContext(sampleRate int, channelCount int, format mux.Format, bufferSizeInBytes int) (*context, chan struct{}, error) {
+	ready := make(chan struct{})
+
+	class := js.Global().Get("AudioContext")
+	if !class.Truthy() {
+		class = js.Global().Get("webkitAudioContext")
+	}
+	if !class.Truthy() {
+		return nil, nil, errors.New("oto: AudioContext or webkitAudioContext was not found")
+	}
+	options := js.Global().Get("Object").New()
+	options.Set("sampleRate", sampleRate)
+
+	d := &context{
+		audioContext: class.New(options),
+		mux:          mux.New(sampleRate, channelCount, format),
+	}
+
+	if bufferSizeInBytes == 0 {
+		// 4096 was not great at least on Safari 15.
+		bufferSizeInBytes = 8192 * channelCount
+	}
+
+	buf32 := make([]float32, bufferSizeInBytes/4)
+	chBuf32 := make([][]float32, channelCount)
+	for i := range chBuf32 {
+		chBuf32[i] = make([]float32, len(buf32)/channelCount)
+	}
+
+	// TODO: Use AudioWorklet if available.
+	sp := d.audioContext.Call("createScriptProcessor", bufferSizeInBytes/4/channelCount, 0, channelCount)
+	f := js.FuncOf(func(this js.Value, arguments []js.Value) any {
+		d.mux.ReadFloat32s(buf32)
+		for i := 0; i < channelCount; i++ {
+			for j := range chBuf32[i] {
+				chBuf32[i][j] = buf32[j*channelCount+i]
+			}
+		}
+
+		buf := arguments[0].Get("outputBuffer")
+		if buf.Get("copyToChannel").Truthy() {
+			for i := 0; i < channelCount; i++ {
+				buf.Call("copyToChannel", float32SliceToTypedArray(chBuf32[i]), i, 0)
+			}
+		} else {
+			// copyToChannel is not defined on Safari 11.
+			for i := 0; i < channelCount; i++ {
+				buf.Call("getChannelData", i).Call("set", float32SliceToTypedArray(chBuf32[i]))
+			}
+		}
+		return nil
+	})
+	sp.Call("addEventListener", "audioprocess", f)
+	d.scriptProcessor = sp
+	d.scriptProcessorCallback = f
+
+	sp.Call("connect", d.audioContext.Get("destination"))
+
+	setCallback := func(event string) js.Func {
+		var f js.Func
+		f = js.FuncOf(func(this js.Value, arguments []js.Value) any {
+			if !d.ready {
+				d.audioContext.Call("resume")
+				d.ready = true
+				close(ready)
+			}
+			js.Global().Get("document").Call("removeEventListener", event, f)
+			return nil
+		})
+		js.Global().Get("document").Call("addEventListener", event, f)
+		d.callbacks[event] = f
+		return f
+	}
+
+	// Browsers require user interaction to start the audio.
+	// https://developers.google.com/web/updates/2017/09/autoplay-policy-changes#webaudio
+	d.callbacks = map[string]js.Func{}
+	setCallback("touchend")
+	setCallback("keyup")
+	setCallback("mouseup")
+
+	return d, ready, nil
+}
+
+func (c *context) Suspend() error {
+	c.audioContext.Call("suspend")
+	return nil
+}
+
+func (c *context) Resume() error {
+	c.audioContext.Call("resume")
+	return nil
+}
+
+func (c *context) Err() error {
+	return nil
+}
+
+func float32SliceToTypedArray(s []float32) js.Value {
+	bs := unsafe.Slice((*byte)(unsafe.Pointer(&s[0])), len(s)*4)
+	a := js.Global().Get("Uint8Array").New(len(bs))
+	js.CopyBytesToJS(a, bs)
+	runtime.KeepAlive(s)
+	buf := a.Get("buffer")
+	return js.Global().Get("Float32Array").New(buf, a.Get("byteOffset"), a.Get("byteLength").Int()/4)
+}

vendor/github.com/ebitengine/oto/v3/driver_macos.go

@@ -0,0 +1,78 @@
+// Copyright 2021 The Oto Authors
+//
+// Licensed under the Apache License, Version 2.0 (the "License");
+// you may not use this file except in compliance with the License.
+// You may obtain a copy of the License at
+//
+//     http://www.apache.org/licenses/LICENSE-2.0
+//
+// Unless required by applicable law or agreed to in writing, software
+// distributed under the License is distributed on an "AS IS" BASIS,
+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+// See the License for the specific language governing permissions and
+// limitations under the License.
+
+//go:build darwin && !ios
+
+package oto
+
+import (
+	"unsafe"
+
+	"github.com/ebitengine/purego"
+	"github.com/ebitengine/purego/objc"
+)
+
+const defaultOneBufferSizeInBytes = 2048
+
+// setNotificationHandler sets a handler for sleep/wake notifications.
+func setNotificationHandler() error {
+	appkit, err := purego.Dlopen("/System/Library/Frameworks/AppKit.framework/Versions/Current/AppKit", purego.RTLD_GLOBAL)
+	if err != nil {
+		return err
+	}
+
+	// Create the Observer object
+	var class objc.Class
+	class, err = objc.RegisterClass("OtoNotificationObserver", objc.GetClass("NSObject"), nil, nil,
+		[]objc.MethodDef{
+			{
+				Cmd: objc.RegisterName("receiveSleepNote:"),
+				Fn:  setGlobalPause,
+			},
+			{
+				Cmd: objc.RegisterName("receiveWakeNote:"),
+				Fn:  setGlobalResume,
+			},
+		})
+
+	observer := objc.ID(class).Send(objc.RegisterName("new"))
+
+	notificationCenter := objc.ID(objc.GetClass("NSWorkspace")).Send(objc.RegisterName("sharedWorkspace")).Send(objc.RegisterName("notificationCenter"))
+
+	// Dlsym returns a pointer to the object so dereference it
+	s, err := purego.Dlsym(appkit, "NSWorkspaceWillSleepNotification")
+	if err != nil {
+		return err
+	}
+
+	notificationCenter.Send(objc.RegisterName("addObserver:selector:name:object:"),
+		observer,
+		objc.RegisterName("receiveSleepNote:"),
+		*(*uintptr)(unsafe.Pointer(s)),
+		0,
+	)
+
+	s, err = purego.Dlsym(appkit, "NSWorkspaceDidWakeNotification")
+	if err != nil {
+		return err
+	}
+
+	notificationCenter.Send(objc.RegisterName("addObserver:selector:name:object:"),
+		observer,
+		objc.RegisterName("receiveWakeNote:"),
+		*(*uintptr)(unsafe.Pointer(s)),
+		0,
+	)
+	return nil
+}

vendor/github.com/ebitengine/oto/v3/driver_nintendosdk.cpp

@@ -0,0 +1,25 @@
+// Copyright 2022 The Oto Authors
+//
+// Licensed under the Apache License, Version 2.0 (the "License");
+// you may not use this file except in compliance with the License.
+// You may obtain a copy of the License at
+//
+//     http://www.apache.org/licenses/LICENSE-2.0
+//
+// Unless required by applicable law or agreed to in writing, software
+// distributed under the License is distributed on an "AS IS" BASIS,
+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+// See the License for the specific language governing permissions and
+// limitations under the License.
+
+//go:build nintendosdk
+
+// The actual implementaiton will be provided by -overlay.
+
+#include <cstddef>
+
+typedef void (*oto_OnReadCallbackType)(float *buf, size_t length);
+
+extern "C" void oto_OpenAudio(int sample_rate, int channel_num,
+                              oto_OnReadCallbackType on_read_callback,
+                              int buffer_size_in_bytes) {}

vendor/github.com/ebitengine/oto/v3/driver_nintendosdk.go

@@ -0,0 +1,74 @@
+// Copyright 2022 The Oto Authors
+//
+// Licensed under the Apache License, Version 2.0 (the "License");
+// you may not use this file except in compliance with the License.
+// You may obtain a copy of the License at
+//
+//     http://www.apache.org/licenses/LICENSE-2.0
+//
+// Unless required by applicable law or agreed to in writing, software
+// distributed under the License is distributed on an "AS IS" BASIS,
+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+// See the License for the specific language governing permissions and
+// limitations under the License.
+
+//go:build nintendosdk
+
+package oto
+
+// #cgo !darwin LDFLAGS: -Wl,-unresolved-symbols=ignore-all
+// #cgo darwin LDFLAGS: -Wl,-undefined,dynamic_lookup
+//
+// typedef void (*oto_OnReadCallbackType)(float* buf, size_t length);
+//
+// void oto_OpenAudio(int sample_rate, int channel_num, oto_OnReadCallbackType on_read_callback, int buffer_size_in_bytes);
+//
+// void oto_OnReadCallback(float* buf, size_t length);
+// static void oto_OpenProxy(int sample_rate, int channel_num, int buffer_size_in_bytes) {
+//   oto_OpenAudio(sample_rate, channel_num, oto_OnReadCallback, buffer_size_in_bytes);
+// }
+import "C"
+
+import (
+	"unsafe"
+
+	"github.com/ebitengine/oto/v3/internal/mux"
+)
+
+//export oto_OnReadCallback
+func oto_OnReadCallback(buf *C.float, length C.size_t) {
+	theContext.mux.ReadFloat32s(unsafe.Slice((*float32)(unsafe.Pointer(buf)), length))
+}
+
+type context struct {
+	mux *mux.Mux
+}
+
+var theContext *context
+
+func newContext(sampleRate int, channelCount int, format mux.Format, bufferSizeInBytes int) (*context, chan struct{}, error) {
+	ready := make(chan struct{})
+	close(ready)
+
+	c := &context{
+		mux: mux.New(sampleRate, channelCount, format),
+	}
+	theContext = c
+	C.oto_OpenProxy(C.int(sampleRate), C.int(channelCount), C.int(bufferSizeInBytes))
+
+	return c, ready, nil
+}
+
+func (c *context) Suspend() error {
+	// Do nothing so far.
+	return nil
+}
+
+func (c *context) Resume() error {
+	// Do nothing so far.
+	return nil
+}
+
+func (c *context) Err() error {
+	return nil
+}

vendor/github.com/ebitengine/oto/v3/driver_unix.go

@@ -0,0 +1,278 @@
+// Copyright 2021 The Oto Authors
+//
+// Licensed under the Apache License, Version 2.0 (the "License");
+// you may not use this file except in compliance with the License.
+// You may obtain a copy of the License at
+//
+//     http://www.apache.org/licenses/LICENSE-2.0
+//
+// Unless required by applicable law or agreed to in writing, software
+// distributed under the License is distributed on an "AS IS" BASIS,
+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+// See the License for the specific language governing permissions and
+// limitations under the License.
+
+//go:build !android && !darwin && !js && !windows && !nintendosdk
+
+package oto
+
+// #cgo pkg-config: alsa
+//
+// #include <alsa/asoundlib.h>
+import "C"
+
+import (
+	"fmt"
+	"strings"
+	"sync"
+	"unsafe"
+
+	"github.com/ebitengine/oto/v3/internal/mux"
+)
+
+type context struct {
+	channelCount int
+
+	suspended bool
+
+	handle *C.snd_pcm_t
+
+	cond *sync.Cond
+
+	mux *mux.Mux
+	err atomicError
+
+	ready chan struct{}
+}
+
+var theContext *context
+
+func alsaError(name string, err C.int) error {
+	return fmt.Errorf("oto: ALSA error at %s: %s", name, C.GoString(C.snd_strerror(err)))
+}
+
+func deviceCandidates() []string {
+	const getAllDevices = -1
+
+	cPCMInterfaceName := C.CString("pcm")
+	defer C.free(unsafe.Pointer(cPCMInterfaceName))
+
+	var hints *unsafe.Pointer
+	err := C.snd_device_name_hint(getAllDevices, cPCMInterfaceName, &hints)
+	if err != 0 {
+		return []string{"default", "plug:default"}
+	}
+	defer C.snd_device_name_free_hint(hints)
+
+	var devices []string
+
+	cIoHintName := C.CString("IOID")
+	defer C.free(unsafe.Pointer(cIoHintName))
+	cNameHintName := C.CString("NAME")
+	defer C.free(unsafe.Pointer(cNameHintName))
+
+	for it := hints; *it != nil; it = (*unsafe.Pointer)(unsafe.Pointer(uintptr(unsafe.Pointer(it)) + unsafe.Sizeof(uintptr(0)))) {
+		io := C.snd_device_name_get_hint(*it, cIoHintName)
+		defer func() {
+			if io != nil {
+				C.free(unsafe.Pointer(io))
+			}
+		}()
+		if C.GoString(io) == "Input" {
+			continue
+		}
+
+		name := C.snd_device_name_get_hint(*it, cNameHintName)
+		defer func() {
+			if name != nil {
+				C.free(unsafe.Pointer(name))
+			}
+		}()
+		if name == nil {
+			continue
+		}
+		goName := C.GoString(name)
+		if goName == "null" {
+			continue
+		}
+		if goName == "default" {
+			continue
+		}
+		devices = append(devices, goName)
+	}
+
+	devices = append([]string{"default", "plug:default"}, devices...)
+
+	return devices
+}
+
+func newContext(sampleRate int, channelCount int, format mux.Format, bufferSizeInBytes int) (*context, chan struct{}, error) {
+	c := &context{
+		channelCount: channelCount,
+		cond:         sync.NewCond(&sync.Mutex{}),
+		mux:          mux.New(sampleRate, channelCount, format),
+		ready:        make(chan struct{}),
+	}
+	theContext = c
+
+	go func() {
+		defer close(c.ready)
+
+		// Open a default ALSA audio device for blocking stream playback
+		type openError struct {
+			device string
+			err    C.int
+		}
+		var openErrs []openError
+		var found bool
+
+		for _, name := range deviceCandidates() {
+			cname := C.CString(name)
+			defer C.free(unsafe.Pointer(cname))
+			if err := C.snd_pcm_open(&c.handle, cname, C.SND_PCM_STREAM_PLAYBACK, 0); err < 0 {
+				openErrs = append(openErrs, openError{
+					device: name,
+					err:    err,
+				})
+				continue
+			}
+			found = true
+			break
+		}
+		if !found {
+			var msgs []string
+			for _, e := range openErrs {
+				msgs = append(msgs, fmt.Sprintf("%q: %s", e.device, C.GoString(C.snd_strerror(e.err))))
+			}
+			c.err.TryStore(fmt.Errorf("oto: ALSA error at snd_pcm_open: %s", strings.Join(msgs, ", ")))
+			return
+		}
+
+		// TODO: Should snd_pcm_hw_params_set_periods be called explicitly?
+		const periods = 2
+		var periodSize C.snd_pcm_uframes_t
+		if bufferSizeInBytes != 0 {
+			periodSize = C.snd_pcm_uframes_t(bufferSizeInBytes / (channelCount * 4 * periods))
+		} else {
+			periodSize = C.snd_pcm_uframes_t(1024)
+		}
+		bufferSize := periodSize * periods
+		if err := c.alsaPcmHwParams(sampleRate, channelCount, &bufferSize, &periodSize); err != nil {
+			c.err.TryStore(err)
+			return
+		}
+
+		go func() {
+			buf32 := make([]float32, int(periodSize)*channelCount)
+			for {
+				if !c.readAndWrite(buf32) {
+					return
+				}
+			}
+		}()
+	}()
+
+	return c, c.ready, nil
+}
+
+func (c *context) alsaPcmHwParams(sampleRate, channelCount int, bufferSize, periodSize *C.snd_pcm_uframes_t) error {
+	var params *C.snd_pcm_hw_params_t
+	C.snd_pcm_hw_params_malloc(&params)
+	defer C.free(unsafe.Pointer(params))
+
+	if err := C.snd_pcm_hw_params_any(c.handle, params); err < 0 {
+		return alsaError("snd_pcm_hw_params_any", err)
+	}
+	if err := C.snd_pcm_hw_params_set_access(c.handle, params, C.SND_PCM_ACCESS_RW_INTERLEAVED); err < 0 {
+		return alsaError("snd_pcm_hw_params_set_access", err)
+	}
+	if err := C.snd_pcm_hw_params_set_format(c.handle, params, C.SND_PCM_FORMAT_FLOAT_LE); err < 0 {
+		return alsaError("snd_pcm_hw_params_set_format", err)
+	}
+	if err := C.snd_pcm_hw_params_set_channels(c.handle, params, C.unsigned(channelCount)); err < 0 {
+		return alsaError("snd_pcm_hw_params_set_channels", err)
+	}
+	if err := C.snd_pcm_hw_params_set_rate_resample(c.handle, params, 1); err < 0 {
+		return alsaError("snd_pcm_hw_params_set_rate_resample", err)
+	}
+	sr := C.unsigned(sampleRate)
+	if err := C.snd_pcm_hw_params_set_rate_near(c.handle, params, &sr, nil); err < 0 {
+		return alsaError("snd_pcm_hw_params_set_rate_near", err)
+	}
+	if err := C.snd_pcm_hw_params_set_buffer_size_near(c.handle, params, bufferSize); err < 0 {
+		return alsaError("snd_pcm_hw_params_set_buffer_size_near", err)
+	}
+	if err := C.snd_pcm_hw_params_set_period_size_near(c.handle, params, periodSize, nil); err < 0 {
+		return alsaError("snd_pcm_hw_params_set_period_size_near", err)
+	}
+	if err := C.snd_pcm_hw_params(c.handle, params); err < 0 {
+		return alsaError("snd_pcm_hw_params", err)
+	}
+	return nil
+}
+
+func (c *context) readAndWrite(buf32 []float32) bool {
+	c.cond.L.Lock()
+	defer c.cond.L.Unlock()
+
+	for c.suspended && c.err.Load() == nil {
+		c.cond.Wait()
+	}
+	if c.err.Load() != nil {
+		return false
+	}
+
+	c.mux.ReadFloat32s(buf32)
+
+	for len(buf32) > 0 {
+		n := C.snd_pcm_writei(c.handle, unsafe.Pointer(&buf32[0]), C.snd_pcm_uframes_t(len(buf32)/c.channelCount))
+		if n < 0 {
+			n = C.long(C.snd_pcm_recover(c.handle, C.int(n), 1))
+		}
+		if n < 0 {
+			c.err.TryStore(alsaError("snd_pcm_writei or snd_pcm_recover", C.int(n)))
+			return false
+		}
+		buf32 = buf32[int(n)*c.channelCount:]
+	}
+	return true
+}
+
+func (c *context) Suspend() error {
+	<-c.ready
+
+	c.cond.L.Lock()
+	defer c.cond.L.Unlock()
+
+	if err := c.err.Load(); err != nil {
+		return err.(error)
+	}
+
+	c.suspended = true
+
+	// Do not use snd_pcm_pause as not all devices support this.
+	// Do not use snd_pcm_drop as this might hang (https://github.com/libsdl-org/SDL/blob/a5c610b0a3857d3138f3f3da1f6dc3172c5ea4a8/src/audio/alsa/SDL_alsa_audio.c#L478).
+	return nil
+}
+
+func (c *context) Resume() error {
+	<-c.ready
+
+	c.cond.L.Lock()
+	defer c.cond.L.Unlock()
+
+	if err := c.err.Load(); err != nil {
+		return err.(error)
+	}
+
+	c.suspended = false
+	c.cond.Signal()
+	return nil
+}
+
+func (c *context) Err() error {
+	if err := c.err.Load(); err != nil {
+		return err.(error)
+	}
+	return nil
+}

vendor/github.com/ebitengine/oto/v3/driver_wasapi_windows.go

@@ -0,0 +1,470 @@
+// Copyright 2022 The Oto Authors
+//
+// Licensed under the Apache License, Version 2.0 (the "License");
+// you may not use this file except in compliance with the License.
+// You may obtain a copy of the License at
+//
+//     http://www.apache.org/licenses/LICENSE-2.0
+//
+// Unless required by applicable law or agreed to in writing, software
+// distributed under the License is distributed on an "AS IS" BASIS,
+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+// See the License for the specific language governing permissions and
+// limitations under the License.
+
+package oto
+
+import (
+	"errors"
+	"fmt"
+	"runtime"
+	"sync"
+	"syscall"
+	"time"
+	"unsafe"
+
+	"golang.org/x/sys/windows"
+
+	"github.com/ebitengine/oto/v3/internal/mux"
+)
+
+type comThread struct {
+	funcCh chan func()
+}
+
+func newCOMThread() (*comThread, error) {
+	funcCh := make(chan func())
+	errCh := make(chan error)
+	go func() {
+		runtime.LockOSThread()
+		defer runtime.UnlockOSThread()
+
+		// S_FALSE is returned when CoInitializeEx is nested. This is a successful case.
+		if err := windows.CoInitializeEx(0, windows.COINIT_MULTITHREADED); err != nil && !errors.Is(err, syscall.Errno(windows.S_FALSE)) {
+			errCh <- err
+		}
+		// CoUninitialize should be called even when CoInitializeEx returns S_FALSE.
+		defer windows.CoUninitialize()
+
+		close(errCh)
+
+		for f := range funcCh {
+			f()
+		}
+	}()
+
+	if err := <-errCh; err != nil {
+		return nil, err
+	}
+
+	return &comThread{
+		funcCh: funcCh,
+	}, nil
+}
+
+func (c *comThread) Run(f func()) {
+	ch := make(chan struct{})
+	c.funcCh <- func() {
+		f()
+		close(ch)
+	}
+	<-ch
+}
+
+type wasapiContext struct {
+	sampleRate        int
+	channelCount      int
+	mux               *mux.Mux
+	bufferSizeInBytes int
+
+	comThread     *comThread
+	err           atomicError
+	suspended     bool
+	suspendedCond *sync.Cond
+
+	sampleReadyEvent windows.Handle
+	client           *_IAudioClient2
+	bufferFrames     uint32
+	renderClient     *_IAudioRenderClient
+	currentDeviceID  string
+	enumerator       *_IMMDeviceEnumerator
+
+	buf []float32
+
+	m sync.Mutex
+}
+
+var (
+	errDeviceSwitched     = errors.New("oto: device switched")
+	errFormatNotSupported = errors.New("oto: the specified format is not supported (there is the closest format instead)")
+)
+
+func newWASAPIContext(sampleRate, channelCount int, mux *mux.Mux, bufferSizeInBytes int) (context *wasapiContext, ferr error) {
+	t, err := newCOMThread()
+	if err != nil {
+		return nil, err
+	}
+
+	c := &wasapiContext{
+		sampleRate:        sampleRate,
+		channelCount:      channelCount,
+		mux:               mux,
+		bufferSizeInBytes: bufferSizeInBytes,
+		comThread:         t,
+		suspendedCond:     sync.NewCond(&sync.Mutex{}),
+	}
+
+	ev, err := windows.CreateEventEx(nil, nil, 0, windows.EVENT_ALL_ACCESS)
+	if err != nil {
+		return nil, err
+	}
+	defer func() {
+		if ferr != nil {
+			windows.CloseHandle(ev)
+		}
+	}()
+	c.sampleReadyEvent = ev
+
+	if err := c.start(); err != nil {
+		return nil, err
+	}
+
+	return c, nil
+}
+
+func (c *wasapiContext) isDeviceSwitched() (bool, error) {
+	// If the audio is suspended, do nothing.
+	if c.isSuspended() {
+		return false, nil
+	}
+
+	var switched bool
+	var cerr error
+	c.comThread.Run(func() {
+		device, err := c.enumerator.GetDefaultAudioEndPoint(eRender, eConsole)
+		if err != nil {
+			cerr = err
+			return
+		}
+		defer device.Release()
+
+		id, err := device.GetId()
+		if err != nil {
+			cerr = err
+			return
+		}
+
+		if c.currentDeviceID == id {
+			return
+		}
+		switched = true
+	})
+
+	return switched, cerr
+}
+
+func (c *wasapiContext) start() error {
+	var cerr error
+	c.comThread.Run(func() {
+		if err := c.startOnCOMThread(); err != nil {
+			cerr = err
+			return
+		}
+	})
+	if cerr != nil {
+		return cerr
+	}
+
+	go func() {
+		if err := c.loop(); err != nil {
+			if !errors.Is(err, _AUDCLNT_E_DEVICE_INVALIDATED) && !errors.Is(err, _AUDCLNT_E_RESOURCES_INVALIDATED) && !errors.Is(err, errDeviceSwitched) {
+				c.err.TryStore(err)
+				return
+			}
+
+			if err := c.restart(); err != nil {
+				c.err.TryStore(err)
+				return
+			}
+		}
+	}()
+
+	return nil
+}
+
+func (c *wasapiContext) startOnCOMThread() (ferr error) {
+	if c.enumerator == nil {
+		e, err := _CoCreateInstance(&uuidMMDeviceEnumerator, nil, uint32(_CLSCTX_ALL), &uuidIMMDeviceEnumerator)
+		if err != nil {
+			return err
+		}
+		c.enumerator = (*_IMMDeviceEnumerator)(e)
+		defer func() {
+			if ferr != nil {
+				c.enumerator.Release()
+				c.enumerator = nil
+			}
+		}()
+	}
+
+	device, err := c.enumerator.GetDefaultAudioEndPoint(eRender, eConsole)
+	if err != nil {
+		if errors.Is(err, _E_NOTFOUND) {
+			return errDeviceNotFound
+		}
+		return err
+	}
+	defer device.Release()
+
+	id, err := device.GetId()
+	if err != nil {
+		return err
+	}
+	c.currentDeviceID = id
+
+	if c.client != nil {
+		c.client.Release()
+		c.client = nil
+	}
+
+	client, err := device.Activate(&uuidIAudioClient2, uint32(_CLSCTX_ALL), nil)
+	if err != nil {
+		return err
+	}
+	c.client = (*_IAudioClient2)(client)
+
+	if err := c.client.SetClientProperties(&_AudioClientProperties{
+		cbSize:     uint32(unsafe.Sizeof(_AudioClientProperties{})),
+		bIsOffload: 0,                    // false
+		eCategory:  _AudioCategory_Other, // In the example, AudioCategory_ForegroundOnlyMedia was used, but this value is deprecated.
+	}); err != nil {
+		return err
+	}
+
+	// Check the format is supported by WASAPI.
+	// Stereo with 48000 [Hz] is likely supported, but mono and/or other sample rates are unlikely supported.
+	// Fallback to WinMM in this case anyway.
+	const bitsPerSample = 32
+	nBlockAlign := c.channelCount * bitsPerSample / 8
+	var channelMask uint32
+	switch c.channelCount {
+	case 1:
+		channelMask = _SPEAKER_FRONT_CENTER
+	case 2:
+		channelMask = _SPEAKER_FRONT_LEFT | _SPEAKER_FRONT_RIGHT
+	}
+	f := &_WAVEFORMATEXTENSIBLE{
+		wFormatTag:      _WAVE_FORMAT_EXTENSIBLE,
+		nChannels:       uint16(c.channelCount),
+		nSamplesPerSec:  uint32(c.sampleRate),
+		nAvgBytesPerSec: uint32(c.sampleRate * nBlockAlign),
+		nBlockAlign:     uint16(nBlockAlign),
+		wBitsPerSample:  bitsPerSample,
+		cbSize:          0x16,
+		Samples:         bitsPerSample,
+		dwChannelMask:   channelMask,
+		SubFormat:       _KSDATAFORMAT_SUBTYPE_IEEE_FLOAT,
+	}
+
+	var bufferSizeIn100ns _REFERENCE_TIME
+	if c.bufferSizeInBytes != 0 {
+		bufferSizeInFrames := int64(c.bufferSizeInBytes) / int64(nBlockAlign)
+		bufferSizeIn100ns = _REFERENCE_TIME(1e7 * bufferSizeInFrames / int64(c.sampleRate))
+	} else {
+		// The default buffer size can be too small and might cause glitch noises.
+		// Specify 50[ms] as the buffer size.
+		bufferSizeIn100ns = _REFERENCE_TIME(50 * time.Millisecond / 100)
+	}
+
+	// Even if the sample rate and/or the number of channels are not supported by the audio driver,
+	// AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM should convert the sample rate automatically (#215).
+	if err := c.client.Initialize(_AUDCLNT_SHAREMODE_SHARED,
+		_AUDCLNT_STREAMFLAGS_EVENTCALLBACK|_AUDCLNT_STREAMFLAGS_NOPERSIST|_AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM,
+		bufferSizeIn100ns, 0, f, nil); err != nil {
+		return err
+	}
+
+	frames, err := c.client.GetBufferSize()
+	if err != nil {
+		return err
+	}
+	c.bufferFrames = frames
+
+	if c.renderClient != nil {
+		c.renderClient.Release()
+		c.renderClient = nil
+	}
+
+	renderClient, err := c.client.GetService(&uuidIAudioRenderClient)
+	if err != nil {
+		return err
+	}
+	c.renderClient = (*_IAudioRenderClient)(renderClient)
+
+	if err := c.client.SetEventHandle(c.sampleReadyEvent); err != nil {
+		return err
+	}
+
+	// TODO: Should some errors be allowed? See WASAPIManager.cpp in the official example SimpleWASAPIPlaySound.
+
+	if err := c.client.Start(); err != nil {
+		return err
+	}
+
+	return nil
+}
+
+func (c *wasapiContext) loop() error {
+	runtime.LockOSThread()
+	defer runtime.UnlockOSThread()
+
+	// S_FALSE is returned when CoInitializeEx is nested. This is a successful case.
+	if err := windows.CoInitializeEx(0, windows.COINIT_MULTITHREADED); err != nil && !errors.Is(err, syscall.Errno(windows.S_FALSE)) {
+		_, _ = c.client.Stop()
+		return err
+	}
+	// CoUninitialize should be called even when CoInitializeEx returns S_FALSE.
+	defer windows.CoUninitialize()
+
+	if err := c.loopOnRenderThread(); err != nil {
+		_, _ = c.client.Stop()
+		return err
+	}
+
+	return nil
+}
+
+func (c *wasapiContext) loopOnRenderThread() error {
+	last := time.Now()
+	for {
+		c.suspendedCond.L.Lock()
+		for c.suspended {
+			c.suspendedCond.Wait()
+		}
+		c.suspendedCond.L.Unlock()
+
+		evt, err := windows.WaitForSingleObject(c.sampleReadyEvent, windows.INFINITE)
+		if err != nil {
+			return err
+		}
+		if evt != windows.WAIT_OBJECT_0 {
+			return fmt.Errorf("oto: WaitForSingleObject failed: returned value: %d", evt)
+		}
+
+		if err := c.writeOnRenderThread(); err != nil {
+			return err
+		}
+
+		// Checking the current default audio device might be an expensive operation.
+		// Check this repeatedly but with some time interval.
+		if now := time.Now(); now.Sub(last) >= 500*time.Millisecond {
+			switched, err := c.isDeviceSwitched()
+			if err != nil {
+				return err
+			}
+			if switched {
+				return errDeviceSwitched
+			}
+			last = now
+		}
+	}
+}
+
+func (c *wasapiContext) writeOnRenderThread() error {
+	c.m.Lock()
+	defer c.m.Unlock()
+
+	paddingFrames, err := c.client.GetCurrentPadding()
+	if err != nil {
+		return err
+	}
+
+	frames := c.bufferFrames - paddingFrames
+	if frames <= 0 {
+		return nil
+	}
+
+	// Get the destination buffer.
+	dstBuf, err := c.renderClient.GetBuffer(frames)
+	if err != nil {
+		return err
+	}
+
+	// Calculate the buffer size.
+	if buflen := int(frames) * c.channelCount; cap(c.buf) < buflen {
+		c.buf = make([]float32, buflen)
+	} else {
+		c.buf = c.buf[:buflen]
+	}
+
+	// Read the buffer from the players.
+	c.mux.ReadFloat32s(c.buf)
+
+	// Copy the read buf to the destination buffer.
+	copy(unsafe.Slice((*float32)(unsafe.Pointer(dstBuf)), len(c.buf)), c.buf)
+
+	// Release the buffer.
+	if err := c.renderClient.ReleaseBuffer(frames, 0); err != nil {
+		return err
+	}
+
+	c.buf = c.buf[:0]
+	return nil
+}
+
+func (c *wasapiContext) Suspend() error {
+	c.suspendedCond.L.Lock()
+	c.suspended = true
+	c.suspendedCond.L.Unlock()
+	c.suspendedCond.Signal()
+
+	return nil
+}
+
+func (c *wasapiContext) Resume() error {
+	c.suspendedCond.L.Lock()
+	c.suspended = false
+	c.suspendedCond.L.Unlock()
+	c.suspendedCond.Signal()
+
+	return nil
+}
+
+func (c *wasapiContext) isSuspended() bool {
+	c.suspendedCond.L.Lock()
+	defer c.suspendedCond.L.Unlock()
+	return c.suspended
+}
+
+func (c *wasapiContext) Err() error {
+	return c.err.Load()
+}
+
+func (c *wasapiContext) restart() error {
+	// Probably the driver is missing temporarily e.g. plugging out the headset.
+	// Recreate the device.
+
+retry:
+	c.suspendedCond.L.Lock()
+	for c.suspended {
+		c.suspendedCond.Wait()
+	}
+	c.suspendedCond.L.Unlock()
+
+	if err := c.start(); err != nil {
+		// When a device is switched, the new device might not support the desired format,
+		// or all the audio devices might be disconnected.
+		// Instead of aborting this context, let's wait for the next device switch.
+		if !errors.Is(err, errFormatNotSupported) && !errors.Is(err, errDeviceNotFound) {
+			return err
+		}
+
+		// Just read the buffer and discard it. Then, retry to search the device.
+		var buf32 [4096]float32
+		sleep := time.Duration(float64(time.Second) * float64(len(buf32)) / float64(c.channelCount) / float64(c.sampleRate))
+		c.mux.ReadFloat32s(buf32[:])
+		time.Sleep(sleep)
+		goto retry
+	}
+	return nil
+}

vendor/github.com/ebitengine/oto/v3/driver_windows.go

@@ -0,0 +1,163 @@
+// Copyright 2022 The Oto Authors
+//
+// Licensed under the Apache License, Version 2.0 (the "License");
+// you may not use this file except in compliance with the License.
+// You may obtain a copy of the License at
+//
+//     http://www.apache.org/licenses/LICENSE-2.0
+//
+// Unless required by applicable law or agreed to in writing, software
+// distributed under the License is distributed on an "AS IS" BASIS,
+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+// See the License for the specific language governing permissions and
+// limitations under the License.
+
+package oto
+
+import (
+	"errors"
+	"fmt"
+	"time"
+
+	"github.com/ebitengine/oto/v3/internal/mux"
+)
+
+var errDeviceNotFound = errors.New("oto: device not found")
+
+type context struct {
+	sampleRate   int
+	channelCount int
+
+	mux *mux.Mux
+
+	wasapiContext *wasapiContext
+	winmmContext  *winmmContext
+	nullContext   *nullContext
+
+	ready chan struct{}
+	err   atomicError
+}
+
+func newContext(sampleRate int, channelCount int, format mux.Format, bufferSizeInBytes int) (*context, chan struct{}, error) {
+	ctx := &context{
+		sampleRate:   sampleRate,
+		channelCount: channelCount,
+		mux:          mux.New(sampleRate, channelCount, format),
+		ready:        make(chan struct{}),
+	}
+
+	// Initializing drivers might take some time. Do this asynchronously.
+	go func() {
+		defer close(ctx.ready)
+
+		xc, err0 := newWASAPIContext(sampleRate, channelCount, ctx.mux, bufferSizeInBytes)
+		if err0 == nil {
+			ctx.wasapiContext = xc
+			return
+		}
+
+		wc, err1 := newWinMMContext(sampleRate, channelCount, ctx.mux, bufferSizeInBytes)
+		if err1 == nil {
+			ctx.winmmContext = wc
+			return
+		}
+
+		if errors.Is(err0, errDeviceNotFound) && errors.Is(err1, errDeviceNotFound) {
+			ctx.nullContext = newNullContext(sampleRate, channelCount, ctx.mux)
+			return
+		}
+
+		ctx.err.TryStore(fmt.Errorf("oto: initialization failed: WASAPI: %v, WinMM: %v", err0, err1))
+	}()
+
+	return ctx, ctx.ready, nil
+}
+
+func (c *context) Suspend() error {
+	<-c.ready
+	if c.wasapiContext != nil {
+		return c.wasapiContext.Suspend()
+	}
+	if c.winmmContext != nil {
+		return c.winmmContext.Suspend()
+	}
+	if c.nullContext != nil {
+		return c.nullContext.Suspend()
+	}
+	return nil
+}
+
+func (c *context) Resume() error {
+	<-c.ready
+	if c.wasapiContext != nil {
+		return c.wasapiContext.Resume()
+	}
+	if c.winmmContext != nil {
+		return c.winmmContext.Resume()
+	}
+	if c.nullContext != nil {
+		return c.nullContext.Resume()
+	}
+	return nil
+}
+
+func (c *context) Err() error {
+	if err := c.err.Load(); err != nil {
+		return err
+	}
+
+	select {
+	case <-c.ready:
+	default:
+		return nil
+	}
+
+	if c.wasapiContext != nil {
+		return c.wasapiContext.Err()
+	}
+	if c.winmmContext != nil {
+		return c.winmmContext.Err()
+	}
+	if c.nullContext != nil {
+		return c.nullContext.Err()
+	}
+	return nil
+}
+
+type nullContext struct {
+	suspended bool
+}
+
+func newNullContext(sampleRate int, channelCount int, mux *mux.Mux) *nullContext {
+	c := &nullContext{}
+	go c.loop(sampleRate, channelCount, mux)
+	return c
+}
+
+func (c *nullContext) loop(sampleRate int, channelCount int, mux *mux.Mux) {
+	var buf32 [4096]float32
+	sleep := time.Duration(float64(time.Second) * float64(len(buf32)) / float64(channelCount) / float64(sampleRate))
+	for {
+		if c.suspended {
+			time.Sleep(time.Second)
+			continue
+		}
+
+		mux.ReadFloat32s(buf32[:])
+		time.Sleep(sleep)
+	}
+}
+
+func (c *nullContext) Suspend() error {
+	c.suspended = true
+	return nil
+}
+
+func (c *nullContext) Resume() error {
+	c.suspended = false
+	return nil
+}
+
+func (*nullContext) Err() error {
+	return nil
+}

vendor/github.com/ebitengine/oto/v3/driver_winmm_windows.go

@@ -0,0 +1,297 @@
+// Copyright 2021 The Oto Authors
+//
+// Licensed under the Apache License, Version 2.0 (the "License");
+// you may not use this file except in compliance with the License.
+// You may obtain a copy of the License at
+//
+//     http://www.apache.org/licenses/LICENSE-2.0
+//
+// Unless required by applicable law or agreed to in writing, software
+// distributed under the License is distributed on an "AS IS" BASIS,
+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+// See the License for the specific language governing permissions and
+// limitations under the License.
+
+package oto
+
+import (
+	"errors"
+	"fmt"
+	"sync"
+	"time"
+	"unsafe"
+
+	"golang.org/x/sys/windows"
+
+	"github.com/ebitengine/oto/v3/internal/mux"
+)
+
+// Avoid goroutines on Windows (hajimehoshi/ebiten#1768).
+// Apparently, switching contexts might take longer than other platforms.
+
+const defaultHeaderBufferSize = 4096
+
+type header struct {
+	waveOut uintptr
+	buffer  []float32
+	waveHdr *_WAVEHDR
+}
+
+func newHeader(waveOut uintptr, bufferSizeInBytes int) (*header, error) {
+	h := &header{
+		waveOut: waveOut,
+		buffer:  make([]float32, bufferSizeInBytes/4),
+	}
+	h.waveHdr = &_WAVEHDR{
+		lpData:         uintptr(unsafe.Pointer(&h.buffer[0])),
+		dwBufferLength: uint32(bufferSizeInBytes),
+	}
+	if err := waveOutPrepareHeader(waveOut, h.waveHdr); err != nil {
+		return nil, err
+	}
+	return h, nil
+}
+
+func (h *header) Write(data []float32) error {
+	copy(h.buffer, data)
+	if err := waveOutWrite(h.waveOut, h.waveHdr); err != nil {
+		return err
+	}
+	return nil
+}
+
+func (h *header) IsQueued() bool {
+	return h.waveHdr.dwFlags&_WHDR_INQUEUE != 0
+}
+
+func (h *header) Close() error {
+	return waveOutUnprepareHeader(h.waveOut, h.waveHdr)
+}
+
+type winmmContext struct {
+	sampleRate        int
+	channelCount      int
+	bufferSizeInBytes int
+
+	waveOut uintptr
+	headers []*header
+
+	buf32 []float32
+
+	mux       *mux.Mux
+	err       atomicError
+	loopEndCh chan error
+
+	cond *sync.Cond
+
+	suspended     bool
+	suspendedCond *sync.Cond
+}
+
+var theWinMMContext *winmmContext
+
+func newWinMMContext(sampleRate, channelCount int, mux *mux.Mux, bufferSizeInBytes int) (*winmmContext, error) {
+	// winmm.dll is not available on Xbox.
+	if err := winmm.Load(); err != nil {
+		return nil, fmt.Errorf("oto: loading winmm.dll failed: %w", err)
+	}
+
+	c := &winmmContext{
+		sampleRate:        sampleRate,
+		channelCount:      channelCount,
+		bufferSizeInBytes: bufferSizeInBytes,
+		mux:               mux,
+		cond:              sync.NewCond(&sync.Mutex{}),
+		suspendedCond:     sync.NewCond(&sync.Mutex{}),
+	}
+	theWinMMContext = c
+
+	if err := c.start(); err != nil {
+		return nil, err
+	}
+	return c, nil
+}
+
+func (c *winmmContext) start() error {
+	const bitsPerSample = 32
+	nBlockAlign := c.channelCount * bitsPerSample / 8
+	f := &_WAVEFORMATEX{
+		wFormatTag:      _WAVE_FORMAT_IEEE_FLOAT,
+		nChannels:       uint16(c.channelCount),
+		nSamplesPerSec:  uint32(c.sampleRate),
+		nAvgBytesPerSec: uint32(c.sampleRate * nBlockAlign),
+		nBlockAlign:     uint16(nBlockAlign),
+		wBitsPerSample:  bitsPerSample,
+	}
+
+	// TOOD: What about using an event instead of a callback? PortAudio and other libraries do that.
+	w, err := waveOutOpen(f, waveOutOpenCallback)
+	if errors.Is(err, windows.ERROR_NOT_FOUND) {
+		// This can happen when no device is found (#77).
+		return errDeviceNotFound
+	}
+	if errors.Is(err, _MMSYSERR_BADDEVICEID) {
+		// This can happen when no device is found (hajimehoshi/ebiten#2316).
+		return errDeviceNotFound
+	}
+	if err != nil {
+		return err
+	}
+
+	headerBufferSize := defaultHeaderBufferSize
+	if c.bufferSizeInBytes != 0 {
+		headerBufferSize = c.bufferSizeInBytes
+	}
+
+	c.waveOut = w
+	c.headers = make([]*header, 0, 6)
+	for len(c.headers) < cap(c.headers) {
+		h, err := newHeader(c.waveOut, headerBufferSize)
+		if err != nil {
+			return err
+		}
+		c.headers = append(c.headers, h)
+	}
+
+	c.buf32 = make([]float32, headerBufferSize/4)
+	go c.loop()
+
+	return nil
+}
+
+func (c *winmmContext) Suspend() error {
+	c.suspendedCond.L.Lock()
+	c.suspended = true
+	c.suspendedCond.L.Unlock()
+	c.suspendedCond.Signal()
+
+	return nil
+}
+
+func (c *winmmContext) Resume() (ferr error) {
+	c.suspendedCond.L.Lock()
+	c.suspended = false
+	c.suspendedCond.L.Unlock()
+	c.suspendedCond.Signal()
+
+	return nil
+}
+
+func (c *winmmContext) Err() error {
+	if err := c.err.Load(); err != nil {
+		return err.(error)
+	}
+	return nil
+}
+
+func (c *winmmContext) isHeaderAvailable() bool {
+	for _, h := range c.headers {
+		if !h.IsQueued() {
+			return true
+		}
+	}
+	return false
+}
+
+var waveOutOpenCallback = windows.NewCallback(func(hwo, uMsg, dwInstance, dwParam1, dwParam2 uintptr) uintptr {
+	// Queuing a header in this callback might not work especially when a headset is connected or disconnected.
+	// Just signal the condition vairable and don't do other things.
+	const womDone = 0x3bd
+	if uMsg != womDone {
+		return 0
+	}
+	theWinMMContext.cond.Signal()
+	return 0
+})
+
+func (c *winmmContext) waitUntilHeaderAvailable() bool {
+	c.cond.L.Lock()
+	defer c.cond.L.Unlock()
+
+	for !c.isHeaderAvailable() && c.err.Load() == nil && c.loopEndCh == nil {
+		c.cond.Wait()
+	}
+	return c.err.Load() == nil && c.loopEndCh == nil
+}
+
+func (c *winmmContext) loop() {
+	defer func() {
+		if err := c.closeLoop(); err != nil {
+			c.err.TryStore(err)
+		}
+	}()
+	for {
+		c.suspendedCond.L.Lock()
+		for c.suspended {
+			c.suspendedCond.Wait()
+		}
+		c.suspendedCond.L.Unlock()
+
+		if !c.waitUntilHeaderAvailable() {
+			return
+		}
+		c.appendBuffers()
+	}
+}
+
+func (c *winmmContext) closeLoop() (ferr error) {
+	c.cond.L.Lock()
+	defer c.cond.L.Unlock()
+
+	defer func() {
+		if c.loopEndCh != nil {
+			if ferr != nil {
+				c.loopEndCh <- ferr
+				ferr = nil
+			}
+			close(c.loopEndCh)
+			c.loopEndCh = nil
+		}
+	}()
+
+	for _, h := range c.headers {
+		if err := h.Close(); err != nil {
+			return err
+		}
+	}
+	c.headers = nil
+
+	if err := waveOutClose(c.waveOut); err != nil {
+		return err
+	}
+	c.waveOut = 0
+	return nil
+}
+
+func (c *winmmContext) appendBuffers() {
+	c.cond.L.Lock()
+	defer c.cond.L.Unlock()
+
+	if c.err.Load() != nil {
+		return
+	}
+
+	c.mux.ReadFloat32s(c.buf32)
+
+	for _, h := range c.headers {
+		if h.IsQueued() {
+			continue
+		}
+
+		if err := h.Write(c.buf32); err != nil {
+			switch {
+			case errors.Is(err, _MMSYSERR_NOMEM):
+				continue
+			case errors.Is(err, _MMSYSERR_NODRIVER):
+				sleep := time.Duration(float64(time.Second) * float64(len(c.buf32)) / float64(c.channelCount) / float64(c.sampleRate))
+				time.Sleep(sleep)
+				return
+			case errors.Is(err, windows.ERROR_NOT_FOUND):
+				// This error can happen when e.g. a new HDMI connection is detected (#51).
+				// TODO: Retry later.
+			}
+			c.err.TryStore(fmt.Errorf("oto: Queueing the header failed: %v", err))
+		}
+		return
+	}
+}

vendor/github.com/ebitengine/oto/v3/internal/mux/mux.go

@@ -0,0 +1,560 @@
+// Copyright 2021 The Oto Authors
+//
+// Licensed under the Apache License, Version 2.0 (the "License");
+// you may not use this file except in compliance with the License.
+// You may obtain a copy of the License at
+//
+//     http://www.apache.org/licenses/LICENSE-2.0
+//
+// Unless required by applicable law or agreed to in writing, software
+// distributed under the License is distributed on an "AS IS" BASIS,
+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+// See the License for the specific language governing permissions and
+// limitations under the License.
+
+// Package mux offers APIs for a low-level multiplexer of audio players.
+// Usually you don't have to use this.
+package mux
+
+import (
+	"errors"
+	"fmt"
+	"io"
+	"math"
+	"runtime"
+	"sync"
+	"time"
+)
+
+// Format must sync with oto's Format.
+type Format int
+
+const (
+	FormatFloat32LE Format = iota
+	FormatUnsignedInt8
+	FormatSignedInt16LE
+)
+
+func (f Format) ByteLength() int {
+	switch f {
+	case FormatFloat32LE:
+		return 4
+	case FormatUnsignedInt8:
+		return 1
+	case FormatSignedInt16LE:
+		return 2
+	}
+	panic(fmt.Sprintf("mux: unexpected format: %d", f))
+}
+
+// Mux is a low-level multiplexer of audio players.
+type Mux struct {
+	sampleRate   int
+	channelCount int
+	format       Format
+
+	players map[*playerImpl]struct{}
+	buf     []float32
+	cond    *sync.Cond
+}
+
+// New creates a new Mux.
+func New(sampleRate int, channelCount int, format Format) *Mux {
+	m := &Mux{
+		sampleRate:   sampleRate,
+		channelCount: channelCount,
+		format:       format,
+		cond:         sync.NewCond(&sync.Mutex{}),
+	}
+	go m.loop()
+	return m
+}
+
+func (m *Mux) shouldWait() bool {
+	for p := range m.players {
+		if p.canReadSourceToBuffer() {
+			return false
+		}
+	}
+	return true
+}
+
+func (m *Mux) wait() {
+	m.cond.L.Lock()
+	defer m.cond.L.Unlock()
+
+	for m.shouldWait() {
+		m.cond.Wait()
+	}
+}
+
+func (m *Mux) loop() {
+	var players []*playerImpl
+	for {
+		m.wait()
+
+		m.cond.L.Lock()
+		for i := range players {
+			players[i] = nil
+		}
+		players = players[:0]
+		for p := range m.players {
+			players = append(players, p)
+		}
+		m.cond.L.Unlock()
+
+		allZero := true
+		for _, p := range players {
+			n := p.readSourceToBuffer()
+			if n != 0 {
+				allZero = false
+			}
+		}
+
+		// Sleeping is necessary especially on browsers.
+		// Sometimes a player continues to read 0 bytes from the source and this loop can be a busy loop in such case.
+		if allZero {
+			time.Sleep(time.Millisecond)
+		}
+	}
+}
+
+func (m *Mux) addPlayer(player *playerImpl) {
+	m.cond.L.Lock()
+	defer m.cond.L.Unlock()
+
+	if m.players == nil {
+		m.players = map[*playerImpl]struct{}{}
+	}
+	m.players[player] = struct{}{}
+	m.cond.Signal()
+}
+
+func (m *Mux) removePlayer(player *playerImpl) {
+	m.cond.L.Lock()
+	defer m.cond.L.Unlock()
+
+	delete(m.players, player)
+	m.cond.Signal()
+}
+
+// ReadFloat32s fills buf with the multiplexed data of the players as float32 values.
+func (m *Mux) ReadFloat32s(buf []float32) {
+	m.cond.L.Lock()
+	players := make([]*playerImpl, 0, len(m.players))
+	for p := range m.players {
+		players = append(players, p)
+	}
+	m.cond.L.Unlock()
+
+	for i := range buf {
+		buf[i] = 0
+	}
+	for _, p := range players {
+		p.readBufferAndAdd(buf)
+	}
+	m.cond.Signal()
+}
+
+type Player struct {
+	p *playerImpl
+}
+
+type playerState int
+
+const (
+	playerPaused playerState = iota
+	playerPlay
+	playerClosed
+)
+
+type playerImpl struct {
+	mux        *Mux
+	src        io.Reader
+	prevVolume float64
+	volume     float64
+	err        error
+	state      playerState
+	tmpbuf     []byte
+	buf        []byte
+	eof        bool
+	bufferSize int
+
+	m sync.Mutex
+}
+
+func (m *Mux) NewPlayer(src io.Reader) *Player {
+	pl := &Player{
+		p: &playerImpl{
+			mux:        m,
+			src:        src,
+			prevVolume: 1,
+			volume:     1,
+			bufferSize: m.defaultBufferSize(),
+		},
+	}
+	runtime.SetFinalizer(pl, (*Player).Close)
+	return pl
+}
+
+func (p *Player) Err() error {
+	return p.p.Err()
+}
+
+func (p *playerImpl) Err() error {
+	p.m.Lock()
+	defer p.m.Unlock()
+
+	return p.err
+}
+
+func (p *Player) Play() {
+	p.p.Play()
+}
+
+func (p *playerImpl) Play() {
+	// Goroutines don't work effiently on Windows. Avoid using them (hajimehoshi/ebiten#1768).
+	if runtime.GOOS == "windows" {
+		p.m.Lock()
+		defer p.m.Unlock()
+
+		p.playImpl()
+	} else {
+		ch := make(chan struct{})
+		go func() {
+			p.m.Lock()
+			defer p.m.Unlock()
+
+			close(ch)
+			p.playImpl()
+		}()
+		<-ch
+	}
+}
+
+func (p *Player) SetBufferSize(bufferSize int) {
+	p.p.setBufferSize(bufferSize)
+}
+
+func (p *playerImpl) setBufferSize(bufferSize int) {
+	p.m.Lock()
+	defer p.m.Unlock()
+
+	orig := p.bufferSize
+	p.bufferSize = bufferSize
+	if bufferSize == 0 {
+		p.bufferSize = p.mux.defaultBufferSize()
+	}
+	if orig != p.bufferSize {
+		p.tmpbuf = nil
+	}
+}
+
+func (p *playerImpl) ensureTmpBuf() []byte {
+	if p.tmpbuf == nil {
+		p.tmpbuf = make([]byte, p.bufferSize)
+	}
+	return p.tmpbuf
+}
+
+// read reads the source to buf.
+// read unlocks the mutex temporarily and locks when reading finishes.
+// This avoids locking during an external function call Read (#188).
+//
+// When read is called, the mutex m must be locked.
+func (p *playerImpl) read(buf []byte) (int, error) {
+	p.m.Unlock()
+	defer p.m.Lock()
+	return p.src.Read(buf)
+}
+
+// addToPlayers adds p to the players set.
+//
+// When addToPlayers is called, the mutex m must be locked.
+func (p *playerImpl) addToPlayers() {
+	p.m.Unlock()
+	defer p.m.Lock()
+	p.mux.addPlayer(p)
+}
+
+// removeFromPlayers removes p from the players set.
+//
+// When removeFromPlayers is called, the mutex m must be locked.
+func (p *playerImpl) removeFromPlayers() {
+	p.m.Unlock()
+	defer p.m.Lock()
+	p.mux.removePlayer(p)
+}
+
+func (p *playerImpl) playImpl() {
+	if p.err != nil {
+		return
+	}
+	if p.state != playerPaused {
+		return
+	}
+	p.state = playerPlay
+
+	if !p.eof {
+		buf := p.ensureTmpBuf()
+		for len(p.buf) < p.bufferSize {
+			n, err := p.read(buf)
+			if err != nil && err != io.EOF {
+				p.setErrorImpl(err)
+				return
+			}
+			p.buf = append(p.buf, buf[:n]...)
+			if err == io.EOF {
+				p.eof = true
+				break
+			}
+		}
+	}
+
+	if p.eof && len(p.buf) == 0 {
+		p.state = playerPaused
+	}
+
+	p.addToPlayers()
+}
+
+func (p *Player) Pause() {
+	p.p.Pause()
+}
+
+func (p *playerImpl) Pause() {
+	p.m.Lock()
+	defer p.m.Unlock()
+
+	if p.state != playerPlay {
+		return
+	}
+	p.state = playerPaused
+}
+
+func (p *Player) Seek(offset int64, whence int) (int64, error) {
+	return p.p.Seek(offset, whence)
+}
+
+func (p *playerImpl) Seek(offset int64, whence int) (int64, error) {
+	p.m.Lock()
+	defer p.m.Unlock()
+
+	// If a player is playing, keep playing even after this seeking.
+	if p.state == playerPlay {
+		defer p.playImpl()
+	}
+
+	// Reset the internal buffer.
+	p.resetImpl()
+
+	// Check if the source implements io.Seeker.
+	s, ok := p.src.(io.Seeker)
+	if !ok {
+		return 0, errors.New("mux: the source must implement io.Seeker")
+	}
+	return s.Seek(offset, whence)
+}
+
+func (p *Player) Reset() {
+	p.p.Reset()
+}
+
+func (p *playerImpl) Reset() {
+	p.m.Lock()
+	defer p.m.Unlock()
+	p.resetImpl()
+}
+
+func (p *playerImpl) resetImpl() {
+	if p.state == playerClosed {
+		return
+	}
+	p.state = playerPaused
+	p.buf = p.buf[:0]
+	p.eof = false
+}
+
+func (p *Player) IsPlaying() bool {
+	return p.p.IsPlaying()
+}
+
+func (p *playerImpl) IsPlaying() bool {
+	p.m.Lock()
+	defer p.m.Unlock()
+	return p.state == playerPlay
+}
+
+func (p *Player) Volume() float64 {
+	return p.p.Volume()
+}
+
+func (p *playerImpl) Volume() float64 {
+	p.m.Lock()
+	defer p.m.Unlock()
+	return p.volume
+}
+
+func (p *Player) SetVolume(volume float64) {
+	p.p.SetVolume(volume)
+}
+
+func (p *playerImpl) SetVolume(volume float64) {
+	p.m.Lock()
+	defer p.m.Unlock()
+	p.volume = volume
+	if p.state != playerPlay {
+		p.prevVolume = volume
+	}
+}
+
+func (p *Player) BufferedSize() int {
+	return p.p.BufferedSize()
+}
+
+func (p *playerImpl) BufferedSize() int {
+	p.m.Lock()
+	defer p.m.Unlock()
+	return len(p.buf)
+}
+
+func (p *Player) Close() error {
+	runtime.SetFinalizer(p, nil)
+	return p.p.Close()
+}
+
+func (p *playerImpl) Close() error {
+	p.m.Lock()
+	defer p.m.Unlock()
+	return p.closeImpl()
+}
+
+func (p *playerImpl) closeImpl() error {
+	p.removeFromPlayers()
+
+	if p.state == playerClosed {
+		return p.err
+	}
+	p.state = playerClosed
+	p.buf = nil
+	return p.err
+}
+
+func (p *playerImpl) readBufferAndAdd(buf []float32) int {
+	p.m.Lock()
+	defer p.m.Unlock()
+
+	if p.state != playerPlay {
+		return 0
+	}
+
+	format := p.mux.format
+	bitDepthInBytes := format.ByteLength()
+	n := len(p.buf) / bitDepthInBytes
+	if n > len(buf) {
+		n = len(buf)
+	}
+
+	prevVolume := float32(p.prevVolume)
+	volume := float32(p.volume)
+
+	channelCount := p.mux.channelCount
+	rateDenom := float32(n / channelCount)
+
+	src := p.buf[:n*bitDepthInBytes]
+
+	for i := 0; i < n; i++ {
+		var v float32
+		switch format {
+		case FormatFloat32LE:
+			v = math.Float32frombits(uint32(src[4*i]) | uint32(src[4*i+1])<<8 | uint32(src[4*i+2])<<16 | uint32(src[4*i+3])<<24)
+		case FormatUnsignedInt8:
+			v8 := src[i]
+			v = float32(v8-(1<<7)) / (1 << 7)
+		case FormatSignedInt16LE:
+			v16 := int16(src[2*i]) | (int16(src[2*i+1]) << 8)
+			v = float32(v16) / (1 << 15)
+		default:
+			panic(fmt.Sprintf("mux: unexpected format: %d", format))
+		}
+		if volume == prevVolume {
+			buf[i] += v * volume
+		} else {
+			rate := float32(i/channelCount) / rateDenom
+			if rate > 1 {
+				rate = 1
+			}
+			buf[i] += v * (volume*rate + prevVolume*(1-rate))
+		}
+	}
+
+	p.prevVolume = p.volume
+
+	copy(p.buf, p.buf[n*bitDepthInBytes:])
+	p.buf = p.buf[:len(p.buf)-n*bitDepthInBytes]
+
+	if p.eof && len(p.buf) == 0 {
+		p.state = playerPaused
+	}
+
+	return n
+}
+
+func (p *playerImpl) canReadSourceToBuffer() bool {
+	p.m.Lock()
+	defer p.m.Unlock()
+
+	if p.eof {
+		return false
+	}
+	return len(p.buf) < p.bufferSize
+}
+
+func (p *playerImpl) readSourceToBuffer() int {
+	p.m.Lock()
+	defer p.m.Unlock()
+
+	if p.err != nil {
+		return 0
+	}
+	if p.state == playerClosed {
+		return 0
+	}
+
+	if len(p.buf) >= p.bufferSize {
+		return 0
+	}
+
+	buf := p.ensureTmpBuf()
+	n, err := p.read(buf)
+
+	if err != nil && err != io.EOF {
+		p.setErrorImpl(err)
+		return 0
+	}
+
+	p.buf = append(p.buf, buf[:n]...)
+	if err == io.EOF {
+		p.eof = true
+		if len(p.buf) == 0 {
+			p.state = playerPaused
+		}
+	}
+	return n
+}
+
+func (p *playerImpl) setErrorImpl(err error) {
+	p.err = err
+	p.closeImpl()
+}
+
+// TODO: The term 'buffer' is confusing. Name each buffer with good terms.
+
+// defaultBufferSize returns the default size of the buffer for the audio source.
+// This buffer is used when unreading on pausing the player.
+func (m *Mux) defaultBufferSize() int {
+	bytesPerSample := m.channelCount * m.format.ByteLength()
+	s := m.sampleRate * bytesPerSample / 2 // 0.5[s]
+	// Align s in multiples of bytes per sample, or a buffer could have extra bytes.
+	return s / bytesPerSample * bytesPerSample
+}

vendor/github.com/ebitengine/oto/v3/internal/oboe/.clang-format

@@ -0,0 +1,2 @@
+DisableFormat: true
+SortIncludes: false

vendor/github.com/ebitengine/oto/v3/internal/oboe/LICENSE-oboe

@@ -0,0 +1,202 @@
+
+                                 Apache License
+                           Version 2.0, January 2004
+                        http://www.apache.org/licenses/
+
+   TERMS AND CONDITIONS FOR USE, REPRODUCTION, AND DISTRIBUTION
+
+   1. Definitions.
+
+      "License" shall mean the terms and conditions for use, reproduction,
+      and distribution as defined by Sections 1 through 9 of this document.
+
+      "Licensor" shall mean the copyright owner or entity authorized by
+      the copyright owner that is granting the License.
+
+      "Legal Entity" shall mean the union of the acting entity and all
+      other entities that control, are controlled by, or are under common
+      control with that entity. For the purposes of this definition,
+      "control" means (i) the power, direct or indirect, to cause the
+      direction or management of such entity, whether by contract or
+      otherwise, or (ii) ownership of fifty percent (50%) or more of the
+      outstanding shares, or (iii) beneficial ownership of such entity.
+
+      "You" (or "Your") shall mean an individual or Legal Entity
+      exercising permissions granted by this License.
+
+      "Source" form shall mean the preferred form for making modifications,
+      including but not limited to software source code, documentation
+      source, and configuration files.
+
+      "Object" form shall mean any form resulting from mechanical
+      transformation or translation of a Source form, including but
+      not limited to compiled object code, generated documentation,
+      and conversions to other media types.
+
+      "Work" shall mean the work of authorship, whether in Source or
+      Object form, made available under the License, as indicated by a
+      copyright notice that is included in or attached to the work
+      (an example is provided in the Appendix below).
+
+      "Derivative Works" shall mean any work, whether in Source or Object
+      form, that is based on (or derived from) the Work and for which the
+      editorial revisions, annotations, elaborations, or other modifications
+      represent, as a whole, an original work of authorship. For the purposes
+      of this License, Derivative Works shall not include works that remain
+      separable from, or merely link (or bind by name) to the interfaces of,
+      the Work and Derivative Works thereof.
+
+      "Contribution" shall mean any work of authorship, including
+      the original version of the Work and any modifications or additions
+      to that Work or Derivative Works thereof, that is intentionally
+      submitted to Licensor for inclusion in the Work by the copyright owner
+      or by an individual or Legal Entity authorized to submit on behalf of
+      the copyright owner. For the purposes of this definition, "submitted"
+      means any form of electronic, verbal, or written communication sent
+      to the Licensor or its representatives, including but not limited to
+      communication on electronic mailing lists, source code control systems,
+      and issue tracking systems that are managed by, or on behalf of, the
+      Licensor for the purpose of discussing and improving the Work, but
+      excluding communication that is conspicuously marked or otherwise
+      designated in writing by the copyright owner as "Not a Contribution."
+
+      "Contributor" shall mean Licensor and any individual or Legal Entity
+      on behalf of whom a Contribution has been received by Licensor and
+      subsequently incorporated within the Work.
+
+   2. Grant of Copyright License. Subject to the terms and conditions of
+      this License, each Contributor hereby grants to You a perpetual,
+      worldwide, non-exclusive, no-charge, royalty-free, irrevocable
+      copyright license to reproduce, prepare Derivative Works of,
+      publicly display, publicly perform, sublicense, and distribute the
+      Work and such Derivative Works in Source or Object form.
+
+   3. Grant of Patent License. Subject to the terms and conditions of
+      this License, each Contributor hereby grants to You a perpetual,
+      worldwide, non-exclusive, no-charge, royalty-free, irrevocable
+      (except as stated in this section) patent license to make, have made,
+      use, offer to sell, sell, import, and otherwise transfer the Work,
+      where such license applies only to those patent claims licensable
+      by such Contributor that are necessarily infringed by their
+      Contribution(s) alone or by combination of their Contribution(s)
+      with the Work to which such Contribution(s) was submitted. If You
+      institute patent litigation against any entity (including a
+      cross-claim or counterclaim in a lawsuit) alleging that the Work
+      or a Contribution incorporated within the Work constitutes direct
+      or contributory patent infringement, then any patent licenses
+      granted to You under this License for that Work shall terminate
+      as of the date such litigation is filed.
+
+   4. Redistribution. You may reproduce and distribute copies of the
+      Work or Derivative Works thereof in any medium, with or without
+      modifications, and in Source or Object form, provided that You
+      meet the following conditions:
+
+      (a) You must give any other recipients of the Work or
+          Derivative Works a copy of this License; and
+
+      (b) You must cause any modified files to carry prominent notices
+          stating that You changed the files; and
+
+      (c) You must retain, in the Source form of any Derivative Works
+          that You distribute, all copyright, patent, trademark, and
+          attribution notices from the Source form of the Work,
+          excluding those notices that do not pertain to any part of
+          the Derivative Works; and
+
+      (d) If the Work includes a "NOTICE" text file as part of its
+          distribution, then any Derivative Works that You distribute must
+          include a readable copy of the attribution notices contained
+          within such NOTICE file, excluding those notices that do not
+          pertain to any part of the Derivative Works, in at least one
+          of the following places: within a NOTICE text file distributed
+          as part of the Derivative Works; within the Source form or
+          documentation, if provided along with the Derivative Works; or,
+          within a display generated by the Derivative Works, if and
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vendor/github.com/ebitengine/oto/v3/internal/oboe/README-oboe.md

@@ -0,0 +1,54 @@
+# Oboe [![Build CI](https://github.com/google/oboe/workflows/Build%20CI/badge.svg)](https://github.com/google/oboe/actions)
+
+[![Introduction to Oboe video](docs/images/getting-started-video.jpg)](https://www.youtube.com/watch?v=csfHAbr5ilI&list=PLWz5rJ2EKKc_duWv9IPNvx9YBudNMmLSa)
+
+Oboe is a C++ library which makes it easy to build high-performance audio apps on Android. It was created primarily to allow developers to target a simplified API that works across multiple API levels back to API level 16 (Jelly Bean).
+
+## Features
+- Compatible with API 16 onwards - runs on 99% of Android devices
+- Chooses the audio API (OpenSL ES on API 16+ or AAudio on API 27+) which will give the best audio performance on the target Android device
+- Automatic latency tuning
+- Modern C++ allowing you to write clean, elegant code
+- Workarounds for some known issues
+- [Used by popular apps and frameworks](https://github.com/google/oboe/wiki/AppsUsingOboe)
+
+## Documentation
+- [Getting Started Guide](docs/GettingStarted.md)
+- [Full Guide to Oboe](docs/FullGuide.md)
+- [API reference](https://google.github.io/oboe)
+- [Tech Notes](docs/notes/)
+- [History of Audio features/bugs by Android version](docs/AndroidAudioHistory.md)
+- [Migration guide for apps using OpenSL ES](docs/OpenSLESMigration.md)
+- [Frequently Asked Questions](docs/FAQ.md) (FAQ)
+- [Our roadmap](https://github.com/google/oboe/milestones) - Vote on a feature/issue by adding a thumbs up to the first comment.
+
+### Community
+- Reddit: [r/androidaudiodev](https://www.reddit.com/r/androidaudiodev/)
+- StackOverflow: [#oboe](https://stackoverflow.com/questions/tagged/oboe)
+
+## Testing
+- [**OboeTester** app for measuring latency, glitches, etc.](apps/OboeTester/docs)
+- [Oboe unit tests](tests)
+
+## Videos
+- [Getting started with Oboe](https://www.youtube.com/playlist?list=PLWz5rJ2EKKc_duWv9IPNvx9YBudNMmLSa)
+- [Low Latency Audio - Because Your Ears Are Worth It](https://www.youtube.com/watch?v=8vOf_fDtur4) (Android Dev Summit '18)
+- [Winning on Android](https://www.youtube.com/watch?v=tWBojmBpS74) - How to optimize an Android audio app. (ADC '18)
+
+## Sample code and apps
+- Sample apps can be found in the [samples directory](samples). 
+- A complete "effects processor" app called FXLab can  be found in the [apps/fxlab folder](apps/fxlab). 
+- Also check out the [Rhythm Game codelab](https://developer.android.com/codelabs/musicalgame-using-oboe?hl=en#0).
+
+### Third party sample code
+- [Ableton Link integration demo](https://github.com/jbloit/AndroidLinkAudio) (author: jbloit)
+
+## Contributing
+We would love to receive your pull requests. Before we can though, please read the [contributing](CONTRIBUTING.md) guidelines.
+
+## Version history
+View the [releases page](../../releases).
+
+## License
+[LICENSE](LICENSE)
+

vendor/github.com/ebitengine/oto/v3/internal/oboe/binding_android.cpp

@@ -0,0 +1,191 @@
+// Copyright 2021 The Oto Authors
+//
+// Licensed under the Apache License, Version 2.0 (the "License");
+// you may not use this file except in compliance with the License.
+// You may obtain a copy of the License at
+//
+//     http://www.apache.org/licenses/LICENSE-2.0
+//
+// Unless required by applicable law or agreed to in writing, software
+// distributed under the License is distributed on an "AS IS" BASIS,
+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+// See the License for the specific language governing permissions and
+// limitations under the License.
+
+#include "binding_android.h"
+
+#include "_cgo_export.h"
+#include "oboe_oboe_Oboe_android.h"
+
+#include <condition_variable>
+#include <memory>
+#include <mutex>
+#include <thread>
+#include <vector>
+
+namespace {
+
+class Player;
+
+class Stream : public oboe::AudioStreamDataCallback {
+public:
+  // GetInstance returns the instance of Stream. Only one Stream object is used
+  // in one process. It is because multiple streams can be problematic in both
+  // AAudio and OpenSL (#1656, #1660).
+  static Stream &GetInstance();
+
+  const char *Play(int sample_rate, int channel_num, int buffer_size_in_bytes);
+  const char *Pause();
+  const char *Resume();
+  const char *Close();
+  const char *AppendBuffer(float *buf, size_t len);
+
+  oboe::DataCallbackResult onAudioReady(oboe::AudioStream *oboe_stream,
+                                        void *audio_data,
+                                        int32_t num_frames) override;
+
+private:
+  Stream();
+  void Loop(int num_frames);
+
+  int sample_rate_ = 0;
+  int channel_num_ = 0;
+
+  std::shared_ptr<oboe::AudioStream> stream_;
+
+  // All the member variables other than the thread must be initialized before
+  // the thread.
+  std::vector<float> buf_;
+  std::mutex mutex_;
+  std::condition_variable cond_;
+  std::unique_ptr<std::thread> thread_;
+};
+
+Stream &Stream::GetInstance() {
+  static Stream *stream = new Stream();
+  return *stream;
+}
+
+const char *Stream::Play(int sample_rate, int channel_num,
+                         int buffer_size_in_bytes) {
+  sample_rate_ = sample_rate;
+  channel_num_ = channel_num;
+
+  if (!stream_) {
+    oboe::AudioStreamBuilder builder;
+    builder.setDirection(oboe::Direction::Output)
+        ->setPerformanceMode(oboe::PerformanceMode::LowLatency)
+        ->setSharingMode(oboe::SharingMode::Shared)
+        ->setFormat(oboe::AudioFormat::Float)
+        ->setChannelCount(channel_num_)
+        ->setSampleRate(sample_rate_)
+        ->setDataCallback(this);
+    if (buffer_size_in_bytes) {
+      int buffer_size_in_frames = buffer_size_in_bytes / channel_num / 4;
+      builder.setBufferCapacityInFrames(buffer_size_in_frames);
+    }
+    oboe::Result result = builder.openStream(stream_);
+    if (result != oboe::Result::OK) {
+      return oboe::convertToText(result);
+    }
+  }
+  if (stream_->getSharingMode() != oboe::SharingMode::Shared) {
+    return "oboe::SharingMode::Shared is not available";
+  }
+
+  int num_frames = stream_->getBufferSizeInFrames();
+  thread_ =
+      std::make_unique<std::thread>([this, num_frames]() { Loop(num_frames); });
+
+  // What if the buffer size is not enough?
+  if (oboe::Result result = stream_->start(); result != oboe::Result::OK) {
+    return oboe::convertToText(result);
+  }
+  return nullptr;
+}
+
+const char *Stream::Pause() {
+  if (!stream_) {
+    return nullptr;
+  }
+  if (oboe::Result result = stream_->pause(); result != oboe::Result::OK) {
+    return oboe::convertToText(result);
+  }
+  return nullptr;
+}
+
+const char *Stream::Resume() {
+  if (!stream_) {
+    return "Play is not called yet at Resume";
+  }
+  if (oboe::Result result = stream_->start(); result != oboe::Result::OK) {
+    return oboe::convertToText(result);
+  }
+  return nullptr;
+}
+
+const char *Stream::Close() {
+  // Nobody calls this so far.
+  if (!stream_) {
+    return nullptr;
+  }
+  if (oboe::Result result = stream_->stop(); result != oboe::Result::OK) {
+    return oboe::convertToText(result);
+  }
+  if (oboe::Result result = stream_->close(); result != oboe::Result::OK) {
+    return oboe::convertToText(result);
+  }
+  stream_.reset();
+  return nullptr;
+}
+
+oboe::DataCallbackResult Stream::onAudioReady(oboe::AudioStream *oboe_stream,
+                                              void *audio_data,
+                                              int32_t num_frames) {
+  size_t num = num_frames * channel_num_;
+  // TODO: Do not use a lock in onAudioReady.
+  // https://google.github.io/oboe/reference/classoboe_1_1_audio_stream_data_callback.html#ad8a3a9f609df5fd3a5d885cbe1b2204d
+  {
+    std::unique_lock<std::mutex> lock{mutex_};
+    cond_.wait(lock, [this, num] { return buf_.size() >= num; });
+    std::copy(buf_.begin(), buf_.begin() + num,
+              reinterpret_cast<float *>(audio_data));
+    buf_.erase(buf_.begin(), buf_.begin() + num);
+    cond_.notify_one();
+  }
+  return oboe::DataCallbackResult::Continue;
+}
+
+Stream::Stream() = default;
+
+void Stream::Loop(int num_frames) {
+  std::vector<float> tmp(num_frames * channel_num_ * 3);
+  for (;;) {
+    {
+      std::unique_lock<std::mutex> lock{mutex_};
+      cond_.wait(lock, [this, &tmp] { return buf_.size() < tmp.size(); });
+    }
+    oto_oboe_read(&tmp[0], tmp.size());
+    {
+      std::lock_guard<std::mutex> lock{mutex_};
+      buf_.insert(buf_.end(), tmp.begin(), tmp.end());
+      cond_.notify_one();
+    }
+  }
+}
+
+} // namespace
+
+extern "C" {
+
+const char *oto_oboe_Play(int sample_rate, int channel_num,
+                          int buffer_size_in_bytes) {
+  return Stream::GetInstance().Play(sample_rate, channel_num,
+                                    buffer_size_in_bytes);
+}
+
+const char *oto_oboe_Suspend() { return Stream::GetInstance().Pause(); }
+
+const char *oto_oboe_Resume() { return Stream::GetInstance().Resume(); }
+
+} // extern "C"

vendor/github.com/ebitengine/oto/v3/internal/oboe/binding_android.go

@@ -0,0 +1,60 @@
+// Copyright 2021 The Oto Authors
+//
+// Licensed under the Apache License, Version 2.0 (the "License");
+// you may not use this file except in compliance with the License.
+// You may obtain a copy of the License at
+//
+//     http://www.apache.org/licenses/LICENSE-2.0
+//
+// Unless required by applicable law or agreed to in writing, software
+// distributed under the License is distributed on an "AS IS" BASIS,
+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+// See the License for the specific language governing permissions and
+// limitations under the License.
+
+package oboe
+
+// Disable AAudio (hajimehoshi/ebiten#1634).
+// AAudio doesn't care about plugging in/out of a headphone.
+// See https://github.com/google/oboe/wiki/TechNote_Disconnect
+
+// #cgo CXXFLAGS: -std=c++17 -DOBOE_ENABLE_AAUDIO=0
+// #cgo LDFLAGS: -llog -lOpenSLES -static-libstdc++
+//
+// #include "binding_android.h"
+import "C"
+
+import (
+	"fmt"
+	"unsafe"
+)
+
+var theReadFunc func(buf []float32)
+
+func Play(sampleRate int, channelCount int, readFunc func(buf []float32), bufferSizeInBytes int) error {
+	// Play can invoke the callback. Set the callback before Play.
+	theReadFunc = readFunc
+	if msg := C.oto_oboe_Play(C.int(sampleRate), C.int(channelCount), C.int(bufferSizeInBytes)); msg != nil {
+		return fmt.Errorf("oboe: Play failed: %s", C.GoString(msg))
+	}
+	return nil
+}
+
+func Suspend() error {
+	if msg := C.oto_oboe_Suspend(); msg != nil {
+		return fmt.Errorf("oboe: Suspend failed: %s", C.GoString(msg))
+	}
+	return nil
+}
+
+func Resume() error {
+	if msg := C.oto_oboe_Resume(); msg != nil {
+		return fmt.Errorf("oboe: Resume failed: %s", C.GoString(msg))
+	}
+	return nil
+}
+
+//export oto_oboe_read
+func oto_oboe_read(buf *C.float, len C.size_t) {
+	theReadFunc(unsafe.Slice((*float32)(unsafe.Pointer(buf)), len))
+}

vendor/github.com/ebitengine/oto/v3/internal/oboe/binding_android.h

@@ -0,0 +1,37 @@
+// Copyright 2021 The Oto Authors
+//
+// Licensed under the Apache License, Version 2.0 (the "License");
+// you may not use this file except in compliance with the License.
+// You may obtain a copy of the License at
+//
+//     http://www.apache.org/licenses/LICENSE-2.0
+//
+// Unless required by applicable law or agreed to in writing, software
+// distributed under the License is distributed on an "AS IS" BASIS,
+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+// See the License for the specific language governing permissions and
+// limitations under the License.
+
+#ifndef OBOE_ANDROID_H_
+#define OBOE_ANDROID_H_
+
+#include <stdbool.h>
+#include <stdint.h>
+#include <stdlib.h>
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+typedef uintptr_t PlayerID;
+
+const char *oto_oboe_Play(int sample_rate, int channel_num,
+                          int buffer_size_in_bytes);
+const char *oto_oboe_Suspend();
+const char *oto_oboe_Resume();
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif // OBOE_ANDROID_H_

vendor/github.com/ebitengine/oto/v3/internal/oboe/generate.go

@@ -0,0 +1,17 @@
+// Copyright 2021 The Oto Authors
+//
+// Licensed under the Apache License, Version 2.0 (the "License");
+// you may not use this file except in compliance with the License.
+// You may obtain a copy of the License at
+//
+//     http://www.apache.org/licenses/LICENSE-2.0
+//
+// Unless required by applicable law or agreed to in writing, software
+// distributed under the License is distributed on an "AS IS" BASIS,
+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+// See the License for the specific language governing permissions and
+// limitations under the License.
+
+//go:generate go run gen.go
+
+package oboe

vendor/github.com/ebitengine/oto/v3/internal/oboe/oboe_aaudio_AAudioExtensions_android.h

@@ -0,0 +1,179 @@
+/*
+ * Copyright 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef OBOE_AAUDIO_EXTENSIONS_H
+#define OBOE_AAUDIO_EXTENSIONS_H
+
+#include <dlfcn.h>
+#include <stdint.h>
+
+#include <sys/system_properties.h>
+
+#include "oboe_common_OboeDebug_android.h"
+#include "oboe_oboe_Oboe_android.h"
+#include "oboe_aaudio_AAudioLoader_android.h"
+
+namespace oboe {
+
+#define LIB_AAUDIO_NAME          "libaaudio.so"
+#define FUNCTION_IS_MMAP         "AAudioStream_isMMapUsed"
+#define FUNCTION_SET_MMAP_POLICY "AAudio_setMMapPolicy"
+#define FUNCTION_GET_MMAP_POLICY "AAudio_getMMapPolicy"
+
+#define AAUDIO_ERROR_UNAVAILABLE  static_cast<aaudio_result_t>(Result::ErrorUnavailable)
+
+typedef struct AAudioStreamStruct         AAudioStream;
+
+/**
+ * Call some AAudio test routines that are not part of the normal API.
+ */
+class AAudioExtensions {
+public:
+    AAudioExtensions() {
+        int32_t policy = getIntegerProperty("aaudio.mmap_policy", 0);
+        mMMapSupported = isPolicyEnabled(policy);
+
+        policy = getIntegerProperty("aaudio.mmap_exclusive_policy", 0);
+        mMMapExclusiveSupported = isPolicyEnabled(policy);
+    }
+
+    static bool isPolicyEnabled(int32_t policy) {
+        return (policy == AAUDIO_POLICY_AUTO || policy == AAUDIO_POLICY_ALWAYS);
+    }
+
+    static AAudioExtensions &getInstance() {
+        static AAudioExtensions instance;
+        return instance;
+    }
+
+    bool isMMapUsed(oboe::AudioStream *oboeStream) {
+        AAudioStream *aaudioStream = (AAudioStream *) oboeStream->getUnderlyingStream();
+        return isMMapUsed(aaudioStream);
+    }
+
+    bool isMMapUsed(AAudioStream *aaudioStream) {
+        if (loadSymbols()) return false;
+        if (mAAudioStream_isMMap == nullptr) return false;
+        return mAAudioStream_isMMap(aaudioStream);
+    }
+
+    /**
+     * Controls whether the MMAP data path can be selected when opening a stream.
+     * It has no effect after the stream has been opened.
+     * It only affects the application that calls it. Other apps are not affected.
+     *
+     * @param enabled
+     * @return 0 or a negative error code
+     */
+    int32_t setMMapEnabled(bool enabled) {
+        if (loadSymbols()) return AAUDIO_ERROR_UNAVAILABLE;
+        if (mAAudio_setMMapPolicy == nullptr) return false;
+        return mAAudio_setMMapPolicy(enabled ? AAUDIO_POLICY_AUTO : AAUDIO_POLICY_NEVER);
+    }
+
+    bool isMMapEnabled() {
+        if (loadSymbols()) return false;
+        if (mAAudio_getMMapPolicy == nullptr) return false;
+        int32_t policy = mAAudio_getMMapPolicy();
+        return isPolicyEnabled(policy);
+    }
+
+    bool isMMapSupported() {
+        return mMMapSupported;
+    }
+
+    bool isMMapExclusiveSupported() {
+        return mMMapExclusiveSupported;
+    }
+
+private:
+
+    enum {
+        AAUDIO_POLICY_NEVER = 1,
+        AAUDIO_POLICY_AUTO,
+        AAUDIO_POLICY_ALWAYS
+    };
+    typedef int32_t aaudio_policy_t;
+
+    int getIntegerProperty(const char *name, int defaultValue) {
+        int result = defaultValue;
+        char valueText[PROP_VALUE_MAX] = {0};
+        if (__system_property_get(name, valueText) != 0) {
+            result = atoi(valueText);
+        }
+        return result;
+    }
+
+    /**
+     * Load the function pointers.
+     * This can be called multiple times.
+     * It should only be called from one thread.
+     *
+     * @return 0 if successful or negative error.
+     */
+    aaudio_result_t loadSymbols() {
+        if (mAAudio_getMMapPolicy != nullptr) {
+            return 0;
+        }
+
+        AAudioLoader *libLoader = AAudioLoader::getInstance();
+        int openResult = libLoader->open();
+        if (openResult != 0) {
+            LOGD("%s() could not open " LIB_AAUDIO_NAME, __func__);
+            return AAUDIO_ERROR_UNAVAILABLE;
+        }
+
+        void *libHandle = AAudioLoader::getInstance()->getLibHandle();
+        if (libHandle == nullptr) {
+            LOGE("%s() could not find " LIB_AAUDIO_NAME, __func__);
+            return AAUDIO_ERROR_UNAVAILABLE;
+        }
+
+        mAAudioStream_isMMap = (bool (*)(AAudioStream *stream))
+                dlsym(libHandle, FUNCTION_IS_MMAP);
+        if (mAAudioStream_isMMap == nullptr) {
+            LOGI("%s() could not find " FUNCTION_IS_MMAP, __func__);
+            return AAUDIO_ERROR_UNAVAILABLE;
+        }
+
+        mAAudio_setMMapPolicy = (int32_t (*)(aaudio_policy_t policy))
+                dlsym(libHandle, FUNCTION_SET_MMAP_POLICY);
+        if (mAAudio_setMMapPolicy == nullptr) {
+            LOGI("%s() could not find " FUNCTION_SET_MMAP_POLICY, __func__);
+            return AAUDIO_ERROR_UNAVAILABLE;
+        }
+
+        mAAudio_getMMapPolicy = (aaudio_policy_t (*)())
+                dlsym(libHandle, FUNCTION_GET_MMAP_POLICY);
+        if (mAAudio_getMMapPolicy == nullptr) {
+            LOGI("%s() could not find " FUNCTION_GET_MMAP_POLICY, __func__);
+            return AAUDIO_ERROR_UNAVAILABLE;
+        }
+
+        return 0;
+    }
+
+    bool      mMMapSupported = false;
+    bool      mMMapExclusiveSupported = false;
+
+    bool    (*mAAudioStream_isMMap)(AAudioStream *stream) = nullptr;
+    int32_t (*mAAudio_setMMapPolicy)(aaudio_policy_t policy) = nullptr;
+    aaudio_policy_t (*mAAudio_getMMapPolicy)() = nullptr;
+};
+
+} // namespace oboe
+
+#endif //OBOE_AAUDIO_EXTENSIONS_H

vendor/github.com/ebitengine/oto/v3/internal/oboe/oboe_aaudio_AAudioLoader_android.cpp

@@ -0,0 +1,450 @@
+/*
+ * Copyright 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <dlfcn.h>
+#include "oboe_oboe_Utilities_android.h"
+#include "oboe_common_OboeDebug_android.h"
+#include "oboe_aaudio_AAudioLoader_android.h"
+
+#define LIB_AAUDIO_NAME "libaaudio.so"
+
+namespace oboe {
+
+AAudioLoader::~AAudioLoader() {
+    // Issue 360: thread_local variables with non-trivial destructors
+    // will cause segfaults if the containing library is dlclose()ed on
+    // devices running M or newer, or devices before M when using a static STL.
+    // The simple workaround is to not call dlclose.
+    // https://github.com/android/ndk/wiki/Changelog-r22#known-issues
+    //
+    // The libaaudio and libaaudioclient do not use thread_local.
+    // But, to be safe, we should avoid dlclose() if possible.
+    // Because AAudioLoader is a static Singleton, we can safely skip
+    // calling dlclose() without causing a resource leak.
+    LOGI("%s() dlclose(%s) not called, OK", __func__, LIB_AAUDIO_NAME);
+}
+
+AAudioLoader* AAudioLoader::getInstance() {
+    static AAudioLoader instance;
+    return &instance;
+}
+
+int AAudioLoader::open() {
+    if (mLibHandle != nullptr) {
+        return 0;
+    }
+
+    // Use RTLD_NOW to avoid the unpredictable behavior that RTLD_LAZY can cause.
+    // Also resolving all the links now will prevent a run-time penalty later.
+    mLibHandle = dlopen(LIB_AAUDIO_NAME, RTLD_NOW);
+    if (mLibHandle == nullptr) {
+        LOGI("AAudioLoader::open() could not find " LIB_AAUDIO_NAME);
+        return -1; // TODO review return code
+    } else {
+        LOGD("AAudioLoader():  dlopen(%s) returned %p", LIB_AAUDIO_NAME, mLibHandle);
+    }
+
+    // Load all the function pointers.
+    createStreamBuilder        = load_I_PPB("AAudio_createStreamBuilder");
+    builder_openStream         = load_I_PBPPS("AAudioStreamBuilder_openStream");
+
+    builder_setChannelCount    = load_V_PBI("AAudioStreamBuilder_setChannelCount");
+    if (builder_setChannelCount == nullptr) {
+        // Use old deprecated alias if needed.
+        builder_setChannelCount = load_V_PBI("AAudioStreamBuilder_setSamplesPerFrame");
+    }
+
+    builder_setBufferCapacityInFrames = load_V_PBI("AAudioStreamBuilder_setBufferCapacityInFrames");
+    builder_setDeviceId        = load_V_PBI("AAudioStreamBuilder_setDeviceId");
+    builder_setDirection       = load_V_PBI("AAudioStreamBuilder_setDirection");
+    builder_setFormat          = load_V_PBI("AAudioStreamBuilder_setFormat");
+    builder_setFramesPerDataCallback = load_V_PBI("AAudioStreamBuilder_setFramesPerDataCallback");
+    builder_setSharingMode     = load_V_PBI("AAudioStreamBuilder_setSharingMode");
+    builder_setPerformanceMode = load_V_PBI("AAudioStreamBuilder_setPerformanceMode");
+    builder_setSampleRate      = load_V_PBI("AAudioStreamBuilder_setSampleRate");
+
+    if (getSdkVersion() >= __ANDROID_API_P__){
+        builder_setUsage       = load_V_PBI("AAudioStreamBuilder_setUsage");
+        builder_setContentType = load_V_PBI("AAudioStreamBuilder_setContentType");
+        builder_setInputPreset = load_V_PBI("AAudioStreamBuilder_setInputPreset");
+        builder_setSessionId   = load_V_PBI("AAudioStreamBuilder_setSessionId");
+    }
+
+    if (getSdkVersion() >= __ANDROID_API_S__){
+        builder_setPackageName       = load_V_PBCPH("AAudioStreamBuilder_setPackageName");
+        builder_setAttributionTag    = load_V_PBCPH("AAudioStreamBuilder_setAttributionTag");
+    }
+
+    if (getSdkVersion() >= __ANDROID_API_S_V2__) {
+        builder_setChannelMask = load_V_PBU("AAudioStreamBuilder_setChannelMask");
+    }
+
+    builder_delete             = load_I_PB("AAudioStreamBuilder_delete");
+
+
+    builder_setDataCallback    = load_V_PBPDPV("AAudioStreamBuilder_setDataCallback");
+    builder_setErrorCallback   = load_V_PBPEPV("AAudioStreamBuilder_setErrorCallback");
+
+    stream_read                = load_I_PSPVIL("AAudioStream_read");
+
+    stream_write               = load_I_PSCPVIL("AAudioStream_write");
+
+    stream_waitForStateChange  = load_I_PSTPTL("AAudioStream_waitForStateChange");
+
+    stream_getTimestamp        = load_I_PSKPLPL("AAudioStream_getTimestamp");
+
+    stream_getChannelCount     = load_I_PS("AAudioStream_getChannelCount");
+    if (stream_getChannelCount == nullptr) {
+        // Use old alias if needed.
+        stream_getChannelCount    = load_I_PS("AAudioStream_getSamplesPerFrame");
+    }
+
+    stream_close               = load_I_PS("AAudioStream_close");
+
+    stream_getBufferSize       = load_I_PS("AAudioStream_getBufferSizeInFrames");
+    stream_getDeviceId         = load_I_PS("AAudioStream_getDeviceId");
+    stream_getBufferCapacity   = load_I_PS("AAudioStream_getBufferCapacityInFrames");
+    stream_getFormat           = load_F_PS("AAudioStream_getFormat");
+    stream_getFramesPerBurst   = load_I_PS("AAudioStream_getFramesPerBurst");
+    stream_getFramesRead       = load_L_PS("AAudioStream_getFramesRead");
+    stream_getFramesWritten    = load_L_PS("AAudioStream_getFramesWritten");
+    stream_getPerformanceMode  = load_I_PS("AAudioStream_getPerformanceMode");
+    stream_getSampleRate       = load_I_PS("AAudioStream_getSampleRate");
+    stream_getSharingMode      = load_I_PS("AAudioStream_getSharingMode");
+    stream_getState            = load_I_PS("AAudioStream_getState");
+    stream_getXRunCount        = load_I_PS("AAudioStream_getXRunCount");
+
+    stream_requestStart        = load_I_PS("AAudioStream_requestStart");
+    stream_requestPause        = load_I_PS("AAudioStream_requestPause");
+    stream_requestFlush        = load_I_PS("AAudioStream_requestFlush");
+    stream_requestStop         = load_I_PS("AAudioStream_requestStop");
+
+    stream_setBufferSize       = load_I_PSI("AAudioStream_setBufferSizeInFrames");
+
+    convertResultToText        = load_CPH_I("AAudio_convertResultToText");
+
+    if (getSdkVersion() >= __ANDROID_API_P__){
+        stream_getUsage        = load_I_PS("AAudioStream_getUsage");
+        stream_getContentType  = load_I_PS("AAudioStream_getContentType");
+        stream_getInputPreset  = load_I_PS("AAudioStream_getInputPreset");
+        stream_getSessionId    = load_I_PS("AAudioStream_getSessionId");
+    }
+
+    if (getSdkVersion() >= __ANDROID_API_S_V2__) {
+        stream_getChannelMask = load_U_PS("AAudioStream_getChannelMask");
+    }
+    return 0;
+}
+
+static void AAudioLoader_check(void *proc, const char *functionName) {
+    if (proc == nullptr) {
+        LOGW("AAudioLoader could not find %s", functionName);
+    }
+}
+
+AAudioLoader::signature_I_PPB AAudioLoader::load_I_PPB(const char *functionName) {
+    void *proc = dlsym(mLibHandle, functionName);
+    AAudioLoader_check(proc, functionName);
+    return reinterpret_cast<signature_I_PPB>(proc);
+}
+
+AAudioLoader::signature_CPH_I AAudioLoader::load_CPH_I(const char *functionName) {
+    void *proc = dlsym(mLibHandle, functionName);
+    AAudioLoader_check(proc, functionName);
+    return reinterpret_cast<signature_CPH_I>(proc);
+}
+
+AAudioLoader::signature_V_PBI AAudioLoader::load_V_PBI(const char *functionName) {
+    void *proc = dlsym(mLibHandle, functionName);
+    AAudioLoader_check(proc, functionName);
+    return reinterpret_cast<signature_V_PBI>(proc);
+}
+
+AAudioLoader::signature_V_PBCPH AAudioLoader::load_V_PBCPH(const char *functionName) {
+    void *proc = dlsym(mLibHandle, functionName);
+    AAudioLoader_check(proc, functionName);
+    return reinterpret_cast<signature_V_PBCPH>(proc);
+}
+
+AAudioLoader::signature_V_PBPDPV AAudioLoader::load_V_PBPDPV(const char *functionName) {
+    void *proc = dlsym(mLibHandle, functionName);
+    AAudioLoader_check(proc, functionName);
+    return reinterpret_cast<signature_V_PBPDPV>(proc);
+}
+
+AAudioLoader::signature_V_PBPEPV AAudioLoader::load_V_PBPEPV(const char *functionName) {
+    void *proc = dlsym(mLibHandle, functionName);
+    AAudioLoader_check(proc, functionName);
+    return reinterpret_cast<signature_V_PBPEPV>(proc);
+}
+
+AAudioLoader::signature_I_PSI AAudioLoader::load_I_PSI(const char *functionName) {
+    void *proc = dlsym(mLibHandle, functionName);
+    AAudioLoader_check(proc, functionName);
+    return reinterpret_cast<signature_I_PSI>(proc);
+}
+
+AAudioLoader::signature_I_PS AAudioLoader::load_I_PS(const char *functionName) {
+    void *proc = dlsym(mLibHandle, functionName);
+    AAudioLoader_check(proc, functionName);
+    return reinterpret_cast<signature_I_PS>(proc);
+}
+
+AAudioLoader::signature_L_PS AAudioLoader::load_L_PS(const char *functionName) {
+    void *proc = dlsym(mLibHandle, functionName);
+    AAudioLoader_check(proc, functionName);
+    return reinterpret_cast<signature_L_PS>(proc);
+}
+
+AAudioLoader::signature_F_PS AAudioLoader::load_F_PS(const char *functionName) {
+    void *proc = dlsym(mLibHandle, functionName);
+    AAudioLoader_check(proc, functionName);
+    return reinterpret_cast<signature_F_PS>(proc);
+}
+
+AAudioLoader::signature_B_PS AAudioLoader::load_B_PS(const char *functionName) {
+    void *proc = dlsym(mLibHandle, functionName);
+    AAudioLoader_check(proc, functionName);
+    return reinterpret_cast<signature_B_PS>(proc);
+}
+
+AAudioLoader::signature_I_PB AAudioLoader::load_I_PB(const char *functionName) {
+    void *proc = dlsym(mLibHandle, functionName);
+    AAudioLoader_check(proc, functionName);
+    return reinterpret_cast<signature_I_PB>(proc);
+}
+
+AAudioLoader::signature_I_PBPPS AAudioLoader::load_I_PBPPS(const char *functionName) {
+    void *proc = dlsym(mLibHandle, functionName);
+    AAudioLoader_check(proc, functionName);
+    return reinterpret_cast<signature_I_PBPPS>(proc);
+}
+
+AAudioLoader::signature_I_PSCPVIL AAudioLoader::load_I_PSCPVIL(const char *functionName) {
+    void *proc = dlsym(mLibHandle, functionName);
+    AAudioLoader_check(proc, functionName);
+    return reinterpret_cast<signature_I_PSCPVIL>(proc);
+}
+
+AAudioLoader::signature_I_PSPVIL AAudioLoader::load_I_PSPVIL(const char *functionName) {
+    void *proc = dlsym(mLibHandle, functionName);
+    AAudioLoader_check(proc, functionName);
+    return reinterpret_cast<signature_I_PSPVIL>(proc);
+}
+
+AAudioLoader::signature_I_PSTPTL AAudioLoader::load_I_PSTPTL(const char *functionName) {
+    void *proc = dlsym(mLibHandle, functionName);
+    AAudioLoader_check(proc, functionName);
+    return reinterpret_cast<signature_I_PSTPTL>(proc);
+}
+
+AAudioLoader::signature_I_PSKPLPL AAudioLoader::load_I_PSKPLPL(const char *functionName) {
+    void *proc = dlsym(mLibHandle, functionName);
+    AAudioLoader_check(proc, functionName);
+    return reinterpret_cast<signature_I_PSKPLPL>(proc);
+}
+
+AAudioLoader::signature_V_PBU AAudioLoader::load_V_PBU(const char *functionName) {
+    void *proc = dlsym(mLibHandle, functionName);
+    AAudioLoader_check(proc, functionName);
+    return reinterpret_cast<signature_V_PBU>(proc);
+}
+
+AAudioLoader::signature_U_PS AAudioLoader::load_U_PS(const char *functionName) {
+    void *proc = dlsym(mLibHandle, functionName);
+    AAudioLoader_check(proc, functionName);
+    return reinterpret_cast<signature_U_PS>(proc);
+}
+
+// Ensure that all AAudio primitive data types are int32_t
+#define ASSERT_INT32(type) static_assert(std::is_same<int32_t, type>::value, \
+#type" must be int32_t")
+
+// Ensure that all AAudio primitive data types are uint32_t
+#define ASSERT_UINT32(type) static_assert(std::is_same<uint32_t, type>::value, \
+#type" must be uint32_t")
+
+#define ERRMSG "Oboe constants must match AAudio constants."
+
+// These asserts help verify that the Oboe definitions match the equivalent AAudio definitions.
+// This code is in this .cpp file so it only gets tested once.
+#ifdef AAUDIO_AAUDIO_H
+
+    ASSERT_INT32(aaudio_stream_state_t);
+    ASSERT_INT32(aaudio_direction_t);
+    ASSERT_INT32(aaudio_format_t);
+    ASSERT_INT32(aaudio_data_callback_result_t);
+    ASSERT_INT32(aaudio_result_t);
+    ASSERT_INT32(aaudio_sharing_mode_t);
+    ASSERT_INT32(aaudio_performance_mode_t);
+
+    static_assert((int32_t)StreamState::Uninitialized == AAUDIO_STREAM_STATE_UNINITIALIZED, ERRMSG);
+    static_assert((int32_t)StreamState::Unknown == AAUDIO_STREAM_STATE_UNKNOWN, ERRMSG);
+    static_assert((int32_t)StreamState::Open == AAUDIO_STREAM_STATE_OPEN, ERRMSG);
+    static_assert((int32_t)StreamState::Starting == AAUDIO_STREAM_STATE_STARTING, ERRMSG);
+    static_assert((int32_t)StreamState::Started == AAUDIO_STREAM_STATE_STARTED, ERRMSG);
+    static_assert((int32_t)StreamState::Pausing == AAUDIO_STREAM_STATE_PAUSING, ERRMSG);
+    static_assert((int32_t)StreamState::Paused == AAUDIO_STREAM_STATE_PAUSED, ERRMSG);
+    static_assert((int32_t)StreamState::Flushing == AAUDIO_STREAM_STATE_FLUSHING, ERRMSG);
+    static_assert((int32_t)StreamState::Flushed == AAUDIO_STREAM_STATE_FLUSHED, ERRMSG);
+    static_assert((int32_t)StreamState::Stopping == AAUDIO_STREAM_STATE_STOPPING, ERRMSG);
+    static_assert((int32_t)StreamState::Stopped == AAUDIO_STREAM_STATE_STOPPED, ERRMSG);
+    static_assert((int32_t)StreamState::Closing == AAUDIO_STREAM_STATE_CLOSING, ERRMSG);
+    static_assert((int32_t)StreamState::Closed == AAUDIO_STREAM_STATE_CLOSED, ERRMSG);
+    static_assert((int32_t)StreamState::Disconnected == AAUDIO_STREAM_STATE_DISCONNECTED, ERRMSG);
+
+    static_assert((int32_t)Direction::Output == AAUDIO_DIRECTION_OUTPUT, ERRMSG);
+    static_assert((int32_t)Direction::Input == AAUDIO_DIRECTION_INPUT, ERRMSG);
+
+    static_assert((int32_t)AudioFormat::Invalid == AAUDIO_FORMAT_INVALID, ERRMSG);
+    static_assert((int32_t)AudioFormat::Unspecified == AAUDIO_FORMAT_UNSPECIFIED, ERRMSG);
+    static_assert((int32_t)AudioFormat::I16 == AAUDIO_FORMAT_PCM_I16, ERRMSG);
+    static_assert((int32_t)AudioFormat::Float == AAUDIO_FORMAT_PCM_FLOAT, ERRMSG);
+
+    static_assert((int32_t)DataCallbackResult::Continue == AAUDIO_CALLBACK_RESULT_CONTINUE, ERRMSG);
+    static_assert((int32_t)DataCallbackResult::Stop == AAUDIO_CALLBACK_RESULT_STOP, ERRMSG);
+
+    static_assert((int32_t)Result::OK == AAUDIO_OK, ERRMSG);
+    static_assert((int32_t)Result::ErrorBase == AAUDIO_ERROR_BASE, ERRMSG);
+    static_assert((int32_t)Result::ErrorDisconnected == AAUDIO_ERROR_DISCONNECTED, ERRMSG);
+    static_assert((int32_t)Result::ErrorIllegalArgument == AAUDIO_ERROR_ILLEGAL_ARGUMENT, ERRMSG);
+    static_assert((int32_t)Result::ErrorInternal == AAUDIO_ERROR_INTERNAL, ERRMSG);
+    static_assert((int32_t)Result::ErrorInvalidState == AAUDIO_ERROR_INVALID_STATE, ERRMSG);
+    static_assert((int32_t)Result::ErrorInvalidHandle == AAUDIO_ERROR_INVALID_HANDLE, ERRMSG);
+    static_assert((int32_t)Result::ErrorUnimplemented == AAUDIO_ERROR_UNIMPLEMENTED, ERRMSG);
+    static_assert((int32_t)Result::ErrorUnavailable == AAUDIO_ERROR_UNAVAILABLE, ERRMSG);
+    static_assert((int32_t)Result::ErrorNoFreeHandles == AAUDIO_ERROR_NO_FREE_HANDLES, ERRMSG);
+    static_assert((int32_t)Result::ErrorNoMemory == AAUDIO_ERROR_NO_MEMORY, ERRMSG);
+    static_assert((int32_t)Result::ErrorNull == AAUDIO_ERROR_NULL, ERRMSG);
+    static_assert((int32_t)Result::ErrorTimeout == AAUDIO_ERROR_TIMEOUT, ERRMSG);
+    static_assert((int32_t)Result::ErrorWouldBlock == AAUDIO_ERROR_WOULD_BLOCK, ERRMSG);
+    static_assert((int32_t)Result::ErrorInvalidFormat == AAUDIO_ERROR_INVALID_FORMAT, ERRMSG);
+    static_assert((int32_t)Result::ErrorOutOfRange == AAUDIO_ERROR_OUT_OF_RANGE, ERRMSG);
+    static_assert((int32_t)Result::ErrorNoService == AAUDIO_ERROR_NO_SERVICE, ERRMSG);
+    static_assert((int32_t)Result::ErrorInvalidRate == AAUDIO_ERROR_INVALID_RATE, ERRMSG);
+
+    static_assert((int32_t)SharingMode::Exclusive == AAUDIO_SHARING_MODE_EXCLUSIVE, ERRMSG);
+    static_assert((int32_t)SharingMode::Shared == AAUDIO_SHARING_MODE_SHARED, ERRMSG);
+
+    static_assert((int32_t)PerformanceMode::None == AAUDIO_PERFORMANCE_MODE_NONE, ERRMSG);
+    static_assert((int32_t)PerformanceMode::PowerSaving
+            == AAUDIO_PERFORMANCE_MODE_POWER_SAVING, ERRMSG);
+    static_assert((int32_t)PerformanceMode::LowLatency
+            == AAUDIO_PERFORMANCE_MODE_LOW_LATENCY, ERRMSG);
+
+// The aaudio_ usage, content and input_preset types were added in NDK 17,
+// which is the first version to support Android Pie (API 28).
+#if __NDK_MAJOR__ >= 17
+
+    ASSERT_INT32(aaudio_usage_t);
+    ASSERT_INT32(aaudio_content_type_t);
+    ASSERT_INT32(aaudio_input_preset_t);
+
+    static_assert((int32_t)Usage::Media == AAUDIO_USAGE_MEDIA, ERRMSG);
+    static_assert((int32_t)Usage::VoiceCommunication == AAUDIO_USAGE_VOICE_COMMUNICATION, ERRMSG);
+    static_assert((int32_t)Usage::VoiceCommunicationSignalling
+            == AAUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING, ERRMSG);
+    static_assert((int32_t)Usage::Alarm == AAUDIO_USAGE_ALARM, ERRMSG);
+    static_assert((int32_t)Usage::Notification == AAUDIO_USAGE_NOTIFICATION, ERRMSG);
+    static_assert((int32_t)Usage::NotificationRingtone == AAUDIO_USAGE_NOTIFICATION_RINGTONE, ERRMSG);
+    static_assert((int32_t)Usage::NotificationEvent == AAUDIO_USAGE_NOTIFICATION_EVENT, ERRMSG);
+    static_assert((int32_t)Usage::AssistanceAccessibility == AAUDIO_USAGE_ASSISTANCE_ACCESSIBILITY, ERRMSG);
+    static_assert((int32_t)Usage::AssistanceNavigationGuidance
+            == AAUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE, ERRMSG);
+    static_assert((int32_t)Usage::AssistanceSonification == AAUDIO_USAGE_ASSISTANCE_SONIFICATION, ERRMSG);
+    static_assert((int32_t)Usage::Game == AAUDIO_USAGE_GAME, ERRMSG);
+    static_assert((int32_t)Usage::Assistant == AAUDIO_USAGE_ASSISTANT, ERRMSG);
+
+    static_assert((int32_t)ContentType::Speech == AAUDIO_CONTENT_TYPE_SPEECH, ERRMSG);
+    static_assert((int32_t)ContentType::Music == AAUDIO_CONTENT_TYPE_MUSIC, ERRMSG);
+    static_assert((int32_t)ContentType::Movie == AAUDIO_CONTENT_TYPE_MOVIE, ERRMSG);
+    static_assert((int32_t)ContentType::Sonification == AAUDIO_CONTENT_TYPE_SONIFICATION, ERRMSG);
+
+    static_assert((int32_t)InputPreset::Generic == AAUDIO_INPUT_PRESET_GENERIC, ERRMSG);
+    static_assert((int32_t)InputPreset::Camcorder == AAUDIO_INPUT_PRESET_CAMCORDER, ERRMSG);
+    static_assert((int32_t)InputPreset::VoiceRecognition == AAUDIO_INPUT_PRESET_VOICE_RECOGNITION, ERRMSG);
+    static_assert((int32_t)InputPreset::VoiceCommunication
+            == AAUDIO_INPUT_PRESET_VOICE_COMMUNICATION, ERRMSG);
+    static_assert((int32_t)InputPreset::Unprocessed == AAUDIO_INPUT_PRESET_UNPROCESSED, ERRMSG);
+
+    static_assert((int32_t)SessionId::None == AAUDIO_SESSION_ID_NONE, ERRMSG);
+    static_assert((int32_t)SessionId::Allocate == AAUDIO_SESSION_ID_ALLOCATE, ERRMSG);
+
+#endif // __NDK_MAJOR__ >= 17
+
+// The aaudio channel masks were added in NDK 24,
+// which is the first version to support Android SC_V2 (API 32).
+#if __NDK_MAJOR__ >= 24
+
+    ASSERT_UINT32(aaudio_channel_mask_t);
+
+    static_assert((uint32_t)ChannelMask::FrontLeft == AAUDIO_CHANNEL_FRONT_LEFT, ERRMSG);
+    static_assert((uint32_t)ChannelMask::FrontRight == AAUDIO_CHANNEL_FRONT_RIGHT, ERRMSG);
+    static_assert((uint32_t)ChannelMask::FrontCenter == AAUDIO_CHANNEL_FRONT_CENTER, ERRMSG);
+    static_assert((uint32_t)ChannelMask::LowFrequency == AAUDIO_CHANNEL_LOW_FREQUENCY, ERRMSG);
+    static_assert((uint32_t)ChannelMask::BackLeft == AAUDIO_CHANNEL_BACK_LEFT, ERRMSG);
+    static_assert((uint32_t)ChannelMask::BackRight == AAUDIO_CHANNEL_BACK_RIGHT, ERRMSG);
+    static_assert((uint32_t)ChannelMask::FrontLeftOfCenter == AAUDIO_CHANNEL_FRONT_LEFT_OF_CENTER, ERRMSG);
+    static_assert((uint32_t)ChannelMask::FrontRightOfCenter == AAUDIO_CHANNEL_FRONT_RIGHT_OF_CENTER, ERRMSG);
+    static_assert((uint32_t)ChannelMask::BackCenter == AAUDIO_CHANNEL_BACK_CENTER, ERRMSG);
+    static_assert((uint32_t)ChannelMask::SideLeft == AAUDIO_CHANNEL_SIDE_LEFT, ERRMSG);
+    static_assert((uint32_t)ChannelMask::SideRight == AAUDIO_CHANNEL_SIDE_RIGHT, ERRMSG);
+    static_assert((uint32_t)ChannelMask::TopCenter == AAUDIO_CHANNEL_TOP_CENTER, ERRMSG);
+    static_assert((uint32_t)ChannelMask::TopFrontLeft == AAUDIO_CHANNEL_TOP_FRONT_LEFT, ERRMSG);
+    static_assert((uint32_t)ChannelMask::TopFrontCenter == AAUDIO_CHANNEL_TOP_FRONT_CENTER, ERRMSG);
+    static_assert((uint32_t)ChannelMask::TopFrontRight == AAUDIO_CHANNEL_TOP_FRONT_RIGHT, ERRMSG);
+    static_assert((uint32_t)ChannelMask::TopBackLeft == AAUDIO_CHANNEL_TOP_BACK_LEFT, ERRMSG);
+    static_assert((uint32_t)ChannelMask::TopBackCenter == AAUDIO_CHANNEL_TOP_BACK_CENTER, ERRMSG);
+    static_assert((uint32_t)ChannelMask::TopBackRight == AAUDIO_CHANNEL_TOP_BACK_RIGHT, ERRMSG);
+    static_assert((uint32_t)ChannelMask::TopSideLeft == AAUDIO_CHANNEL_TOP_SIDE_LEFT, ERRMSG);
+    static_assert((uint32_t)ChannelMask::TopSideRight == AAUDIO_CHANNEL_TOP_SIDE_RIGHT, ERRMSG);
+    static_assert((uint32_t)ChannelMask::BottomFrontLeft == AAUDIO_CHANNEL_BOTTOM_FRONT_LEFT, ERRMSG);
+    static_assert((uint32_t)ChannelMask::BottomFrontCenter == AAUDIO_CHANNEL_BOTTOM_FRONT_CENTER, ERRMSG);
+    static_assert((uint32_t)ChannelMask::BottomFrontRight == AAUDIO_CHANNEL_BOTTOM_FRONT_RIGHT, ERRMSG);
+    static_assert((uint32_t)ChannelMask::LowFrequency2 == AAUDIO_CHANNEL_LOW_FREQUENCY_2, ERRMSG);
+    static_assert((uint32_t)ChannelMask::FrontWideLeft == AAUDIO_CHANNEL_FRONT_WIDE_LEFT, ERRMSG);
+    static_assert((uint32_t)ChannelMask::FrontWideRight == AAUDIO_CHANNEL_FRONT_WIDE_RIGHT, ERRMSG);
+    static_assert((uint32_t)ChannelMask::Mono == AAUDIO_CHANNEL_MONO, ERRMSG);
+    static_assert((uint32_t)ChannelMask::Stereo == AAUDIO_CHANNEL_STEREO, ERRMSG);
+    static_assert((uint32_t)ChannelMask::CM2Point1 == AAUDIO_CHANNEL_2POINT1, ERRMSG);
+    static_assert((uint32_t)ChannelMask::Tri == AAUDIO_CHANNEL_TRI, ERRMSG);
+    static_assert((uint32_t)ChannelMask::TriBack == AAUDIO_CHANNEL_TRI_BACK, ERRMSG);
+    static_assert((uint32_t)ChannelMask::CM3Point1 == AAUDIO_CHANNEL_3POINT1, ERRMSG);
+    static_assert((uint32_t)ChannelMask::CM2Point0Point2 == AAUDIO_CHANNEL_2POINT0POINT2, ERRMSG);
+    static_assert((uint32_t)ChannelMask::CM2Point1Point2 == AAUDIO_CHANNEL_2POINT1POINT2, ERRMSG);
+    static_assert((uint32_t)ChannelMask::CM3Point0Point2 == AAUDIO_CHANNEL_3POINT0POINT2, ERRMSG);
+    static_assert((uint32_t)ChannelMask::CM3Point1Point2 == AAUDIO_CHANNEL_3POINT1POINT2, ERRMSG);
+    static_assert((uint32_t)ChannelMask::Quad == AAUDIO_CHANNEL_QUAD, ERRMSG);
+    static_assert((uint32_t)ChannelMask::QuadSide == AAUDIO_CHANNEL_QUAD_SIDE, ERRMSG);
+    static_assert((uint32_t)ChannelMask::Surround == AAUDIO_CHANNEL_SURROUND, ERRMSG);
+    static_assert((uint32_t)ChannelMask::Penta == AAUDIO_CHANNEL_PENTA, ERRMSG);
+    static_assert((uint32_t)ChannelMask::CM5Point1 == AAUDIO_CHANNEL_5POINT1, ERRMSG);
+    static_assert((uint32_t)ChannelMask::CM5Point1Side == AAUDIO_CHANNEL_5POINT1_SIDE, ERRMSG);
+    static_assert((uint32_t)ChannelMask::CM6Point1 == AAUDIO_CHANNEL_6POINT1, ERRMSG);
+    static_assert((uint32_t)ChannelMask::CM7Point1 == AAUDIO_CHANNEL_7POINT1, ERRMSG);
+    static_assert((uint32_t)ChannelMask::CM5Point1Point2 == AAUDIO_CHANNEL_5POINT1POINT2, ERRMSG);
+    static_assert((uint32_t)ChannelMask::CM5Point1Point4 == AAUDIO_CHANNEL_5POINT1POINT4, ERRMSG);
+    static_assert((uint32_t)ChannelMask::CM7Point1Point2 == AAUDIO_CHANNEL_7POINT1POINT2, ERRMSG);
+    static_assert((uint32_t)ChannelMask::CM7Point1Point4 == AAUDIO_CHANNEL_7POINT1POINT4, ERRMSG);
+    static_assert((uint32_t)ChannelMask::CM9Point1Point4 == AAUDIO_CHANNEL_9POINT1POINT4, ERRMSG);
+    static_assert((uint32_t)ChannelMask::CM9Point1Point6 == AAUDIO_CHANNEL_9POINT1POINT6, ERRMSG);
+    static_assert((uint32_t)ChannelMask::FrontBack == AAUDIO_CHANNEL_FRONT_BACK, ERRMSG);
+
+#endif
+
+#endif // AAUDIO_AAUDIO_H
+
+} // namespace oboe

vendor/github.com/ebitengine/oto/v3/internal/oboe/oboe_aaudio_AAudioLoader_android.h

@@ -0,0 +1,263 @@
+/*
+ * Copyright 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef OBOE_AAUDIO_LOADER_H_
+#define OBOE_AAUDIO_LOADER_H_
+
+#include <unistd.h>
+#include "oboe_oboe_Definitions_android.h"
+
+// If the NDK is before O then define this in your build
+// so that AAudio.h will not be included.
+#ifdef OBOE_NO_INCLUDE_AAUDIO
+
+// Define missing types from AAudio.h
+typedef int32_t aaudio_stream_state_t;
+typedef int32_t aaudio_direction_t;
+typedef int32_t aaudio_format_t;
+typedef int32_t aaudio_data_callback_result_t;
+typedef int32_t aaudio_result_t;
+typedef int32_t aaudio_sharing_mode_t;
+typedef int32_t aaudio_performance_mode_t;
+
+typedef struct AAudioStreamStruct         AAudioStream;
+typedef struct AAudioStreamBuilderStruct  AAudioStreamBuilder;
+
+typedef aaudio_data_callback_result_t (*AAudioStream_dataCallback)(
+        AAudioStream *stream,
+        void *userData,
+        void *audioData,
+        int32_t numFrames);
+
+typedef void (*AAudioStream_errorCallback)(
+        AAudioStream *stream,
+        void *userData,
+        aaudio_result_t error);
+
+// These were defined in P
+typedef int32_t aaudio_usage_t;
+typedef int32_t aaudio_content_type_t;
+typedef int32_t aaudio_input_preset_t;
+typedef int32_t aaudio_session_id_t;
+
+// There are a few definitions used by Oboe.
+#define AAUDIO_OK                      static_cast<aaudio_result_t>(Result::OK)
+#define AAUDIO_ERROR_TIMEOUT           static_cast<aaudio_result_t>(Result::ErrorTimeout)
+#define AAUDIO_STREAM_STATE_STARTING   static_cast<aaudio_stream_state_t>(StreamState::Starting)
+#define AAUDIO_STREAM_STATE_STARTED    static_cast<aaudio_stream_state_t>(StreamState::Started)
+#else
+#include <aaudio/AAudio.h>
+#endif
+
+#ifndef __NDK_MAJOR__
+#define __NDK_MAJOR__ 0
+#endif
+
+#if __NDK_MAJOR__ < 24
+// Defined in SC_V2
+typedef uint32_t aaudio_channel_mask_t;
+#endif
+
+#ifndef __ANDROID_API_S__
+#define __ANDROID_API_S__ 31
+#endif
+
+#ifndef __ANDROID_API_S_V2__
+#define __ANDROID_API_S_V2__ 32
+#endif
+
+namespace oboe {
+
+/**
+ * The AAudio API was not available in early versions of Android.
+ * To avoid linker errors, we dynamically link with the functions by name using dlsym().
+ * On older versions this linkage will safely fail.
+ */
+class AAudioLoader {
+  public:
+    // Use signatures for common functions.
+    // Key to letter abbreviations.
+    // S = Stream
+    // B = Builder
+    // I = int32_t
+    // L = int64_t
+    // T = sTate
+    // K = clocKid_t
+    // P = Pointer to following data type
+    // C = Const prefix
+    // H = cHar
+    // U = uint32_t
+    typedef int32_t  (*signature_I_PPB)(AAudioStreamBuilder **builder);
+
+    typedef const char * (*signature_CPH_I)(int32_t);
+
+    typedef int32_t (*signature_I_PBPPS)(AAudioStreamBuilder *,
+                                      AAudioStream **stream);  // AAudioStreamBuilder_open()
+
+    typedef int32_t (*signature_I_PB)(AAudioStreamBuilder *);  // AAudioStreamBuilder_delete()
+    // AAudioStreamBuilder_setSampleRate()
+    typedef void    (*signature_V_PBI)(AAudioStreamBuilder *, int32_t);
+
+    // AAudioStreamBuilder_setChannelMask()
+    typedef void    (*signature_V_PBU)(AAudioStreamBuilder *, uint32_t);
+
+    typedef void    (*signature_V_PBCPH)(AAudioStreamBuilder *, const char *);
+
+    typedef int32_t (*signature_I_PS)(AAudioStream *);  // AAudioStream_getSampleRate()
+    typedef int64_t (*signature_L_PS)(AAudioStream *);  // AAudioStream_getFramesRead()
+    // AAudioStream_setBufferSizeInFrames()
+    typedef int32_t (*signature_I_PSI)(AAudioStream *, int32_t);
+
+    typedef void    (*signature_V_PBPDPV)(AAudioStreamBuilder *,
+                                          AAudioStream_dataCallback,
+                                          void *);
+
+    typedef void    (*signature_V_PBPEPV)(AAudioStreamBuilder *,
+                                          AAudioStream_errorCallback,
+                                          void *);
+
+    typedef aaudio_format_t (*signature_F_PS)(AAudioStream *stream);
+
+    typedef int32_t (*signature_I_PSPVIL)(AAudioStream *, void *, int32_t, int64_t);
+    typedef int32_t (*signature_I_PSCPVIL)(AAudioStream *, const void *, int32_t, int64_t);
+
+    typedef int32_t (*signature_I_PSTPTL)(AAudioStream *,
+                                          aaudio_stream_state_t,
+                                          aaudio_stream_state_t *,
+                                          int64_t);
+
+    typedef int32_t (*signature_I_PSKPLPL)(AAudioStream *, clockid_t, int64_t *, int64_t *);
+
+    typedef bool    (*signature_B_PS)(AAudioStream *);
+
+    typedef uint32_t (*signature_U_PS)(AAudioStream *);
+
+    static AAudioLoader* getInstance(); // singleton
+
+    /**
+     * Open the AAudio shared library and load the function pointers.
+     * This can be called multiple times.
+     * It should only be called from one thread.
+     *
+     * The destructor will clean up after the open.
+     *
+     * @return 0 if successful or negative error.
+     */
+    int open();
+
+    void *getLibHandle() const { return mLibHandle; }
+
+    // Function pointers into the AAudio shared library.
+    signature_I_PPB   createStreamBuilder = nullptr;
+
+    signature_I_PBPPS builder_openStream = nullptr;
+
+    signature_V_PBI builder_setBufferCapacityInFrames = nullptr;
+    signature_V_PBI builder_setChannelCount = nullptr;
+    signature_V_PBI builder_setDeviceId = nullptr;
+    signature_V_PBI builder_setDirection = nullptr;
+    signature_V_PBI builder_setFormat = nullptr;
+    signature_V_PBI builder_setFramesPerDataCallback = nullptr;
+    signature_V_PBI builder_setPerformanceMode = nullptr;
+    signature_V_PBI builder_setSampleRate = nullptr;
+    signature_V_PBI builder_setSharingMode = nullptr;
+    signature_V_PBU builder_setChannelMask = nullptr;
+
+    signature_V_PBI builder_setUsage = nullptr;
+    signature_V_PBI builder_setContentType = nullptr;
+    signature_V_PBI builder_setInputPreset = nullptr;
+    signature_V_PBI builder_setSessionId = nullptr;
+
+    signature_V_PBCPH builder_setPackageName = nullptr;
+    signature_V_PBCPH builder_setAttributionTag = nullptr;
+
+    signature_V_PBPDPV  builder_setDataCallback = nullptr;
+    signature_V_PBPEPV  builder_setErrorCallback = nullptr;
+
+    signature_I_PB      builder_delete = nullptr;
+
+    signature_F_PS      stream_getFormat = nullptr;
+
+    signature_I_PSPVIL  stream_read = nullptr;
+    signature_I_PSCPVIL stream_write = nullptr;
+
+    signature_I_PSTPTL  stream_waitForStateChange = nullptr;
+
+    signature_I_PSKPLPL stream_getTimestamp = nullptr;
+
+    signature_I_PS   stream_close = nullptr;
+
+    signature_I_PS   stream_getChannelCount = nullptr;
+    signature_I_PS   stream_getDeviceId = nullptr;
+
+    signature_I_PS   stream_getBufferSize = nullptr;
+    signature_I_PS   stream_getBufferCapacity = nullptr;
+    signature_I_PS   stream_getFramesPerBurst = nullptr;
+    signature_I_PS   stream_getState = nullptr;
+    signature_I_PS   stream_getPerformanceMode = nullptr;
+    signature_I_PS   stream_getSampleRate = nullptr;
+    signature_I_PS   stream_getSharingMode = nullptr;
+    signature_I_PS   stream_getXRunCount = nullptr;
+
+    signature_I_PSI  stream_setBufferSize = nullptr;
+    signature_I_PS   stream_requestStart = nullptr;
+    signature_I_PS   stream_requestPause = nullptr;
+    signature_I_PS   stream_requestFlush = nullptr;
+    signature_I_PS   stream_requestStop = nullptr;
+
+    signature_L_PS   stream_getFramesRead = nullptr;
+    signature_L_PS   stream_getFramesWritten = nullptr;
+
+    signature_CPH_I  convertResultToText = nullptr;
+
+    signature_I_PS   stream_getUsage = nullptr;
+    signature_I_PS   stream_getContentType = nullptr;
+    signature_I_PS   stream_getInputPreset = nullptr;
+    signature_I_PS   stream_getSessionId = nullptr;
+
+    signature_U_PS   stream_getChannelMask = nullptr;
+
+  private:
+    AAudioLoader() {}
+    ~AAudioLoader();
+
+    // Load function pointers for specific signatures.
+    signature_I_PPB     load_I_PPB(const char *name);
+    signature_CPH_I     load_CPH_I(const char *name);
+    signature_V_PBI     load_V_PBI(const char *name);
+    signature_V_PBCPH   load_V_PBCPH(const char *name);
+    signature_V_PBPDPV  load_V_PBPDPV(const char *name);
+    signature_V_PBPEPV  load_V_PBPEPV(const char *name);
+    signature_I_PB      load_I_PB(const char *name);
+    signature_I_PBPPS   load_I_PBPPS(const char *name);
+    signature_I_PS      load_I_PS(const char *name);
+    signature_L_PS      load_L_PS(const char *name);
+    signature_F_PS      load_F_PS(const char *name);
+    signature_B_PS      load_B_PS(const char *name);
+    signature_I_PSI     load_I_PSI(const char *name);
+    signature_I_PSPVIL  load_I_PSPVIL(const char *name);
+    signature_I_PSCPVIL load_I_PSCPVIL(const char *name);
+    signature_I_PSTPTL  load_I_PSTPTL(const char *name);
+    signature_I_PSKPLPL load_I_PSKPLPL(const char *name);
+    signature_V_PBU     load_V_PBU(const char *name);
+    signature_U_PS      load_U_PS(const char *name);
+
+    void *mLibHandle = nullptr;
+};
+
+} // namespace oboe
+
+#endif //OBOE_AAUDIO_LOADER_H_

vendor/github.com/ebitengine/oto/v3/internal/oboe/oboe_aaudio_AudioStreamAAudio_android.cpp

@@ -0,0 +1,729 @@
+/*
+ * Copyright 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <cassert>
+#include <stdint.h>
+#include <stdlib.h>
+
+#include "oboe_aaudio_AAudioLoader_android.h"
+#include "oboe_aaudio_AudioStreamAAudio_android.h"
+#include "oboe_common_AudioClock_android.h"
+#include "oboe_common_OboeDebug_android.h"
+#include "oboe_oboe_Utilities_android.h"
+#include "oboe_aaudio_AAudioExtensions_android.h"
+
+#ifdef __ANDROID__
+#include <sys/system_properties.h>
+#include "oboe_common_QuirksManager_android.h"
+
+#endif
+
+#ifndef OBOE_FIX_FORCE_STARTING_TO_STARTED
+// Workaround state problems in AAudio
+// TODO Which versions does this occur in? Verify fixed in Q.
+#define OBOE_FIX_FORCE_STARTING_TO_STARTED 1
+#endif // OBOE_FIX_FORCE_STARTING_TO_STARTED
+
+using namespace oboe;
+AAudioLoader *AudioStreamAAudio::mLibLoader = nullptr;
+
+// 'C' wrapper for the data callback method
+static aaudio_data_callback_result_t oboe_aaudio_data_callback_proc(
+        AAudioStream *stream,
+        void *userData,
+        void *audioData,
+        int32_t numFrames) {
+
+    AudioStreamAAudio *oboeStream = reinterpret_cast<AudioStreamAAudio*>(userData);
+    if (oboeStream != nullptr) {
+        return static_cast<aaudio_data_callback_result_t>(
+                oboeStream->callOnAudioReady(stream, audioData, numFrames));
+
+    } else {
+        return static_cast<aaudio_data_callback_result_t>(DataCallbackResult::Stop);
+    }
+}
+
+// This runs in its own thread.
+// Only one of these threads will be launched from internalErrorCallback().
+// It calls app error callbacks from a static function in case the stream gets deleted.
+static void oboe_aaudio_error_thread_proc(AudioStreamAAudio *oboeStream,
+                                          Result error) {
+    LOGD("%s(,%d) - entering >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>", __func__, error);
+    AudioStreamErrorCallback *errorCallback = oboeStream->getErrorCallback();
+    if (errorCallback == nullptr) return; // should be impossible
+    bool isErrorHandled = errorCallback->onError(oboeStream, error);
+
+    if (!isErrorHandled) {
+        oboeStream->requestStop();
+        errorCallback->onErrorBeforeClose(oboeStream, error);
+        oboeStream->close();
+        // Warning, oboeStream may get deleted by this callback.
+        errorCallback->onErrorAfterClose(oboeStream, error);
+    }
+    LOGD("%s() - exiting <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<", __func__);
+}
+
+// This runs in its own thread.
+// Only one of these threads will be launched from internalErrorCallback().
+// Prevents deletion of the stream if the app is using AudioStreamBuilder::openSharedStream()
+static void oboe_aaudio_error_thread_proc_shared(std::shared_ptr<AudioStream> sharedStream,
+                                          Result error) {
+    AudioStreamAAudio *oboeStream = reinterpret_cast<AudioStreamAAudio*>(sharedStream.get());
+    oboe_aaudio_error_thread_proc(oboeStream, error);
+}
+
+namespace oboe {
+
+/*
+ * Create a stream that uses Oboe Audio API.
+ */
+AudioStreamAAudio::AudioStreamAAudio(const AudioStreamBuilder &builder)
+    : AudioStream(builder)
+    , mAAudioStream(nullptr) {
+    mCallbackThreadEnabled.store(false);
+    mLibLoader = AAudioLoader::getInstance();
+}
+
+bool AudioStreamAAudio::isSupported() {
+    mLibLoader = AAudioLoader::getInstance();
+    int openResult = mLibLoader->open();
+    return openResult == 0;
+}
+
+// Static method for the error callback.
+// We use a method so we can access protected methods on the stream.
+// Launch a thread to handle the error.
+// That other thread can safely stop, close and delete the stream.
+void AudioStreamAAudio::internalErrorCallback(
+        AAudioStream *stream,
+        void *userData,
+        aaudio_result_t error) {
+    oboe::Result oboeResult = static_cast<Result>(error);
+    AudioStreamAAudio *oboeStream = reinterpret_cast<AudioStreamAAudio*>(userData);
+
+    // Coerce the error code if needed to workaround a regression in RQ1A that caused
+    // the wrong code to be passed when headsets plugged in. See b/173928197.
+    if (OboeGlobals::areWorkaroundsEnabled()
+            && getSdkVersion() == __ANDROID_API_R__
+            && oboeResult == oboe::Result::ErrorTimeout) {
+        oboeResult = oboe::Result::ErrorDisconnected;
+        LOGD("%s() ErrorTimeout changed to ErrorDisconnected to fix b/173928197", __func__);
+    }
+
+    oboeStream->mErrorCallbackResult = oboeResult;
+
+    // Prevents deletion of the stream if the app is using AudioStreamBuilder::openStream(shared_ptr)
+    std::shared_ptr<AudioStream> sharedStream = oboeStream->lockWeakThis();
+
+    // These checks should be enough because we assume that the stream close()
+    // will join() any active callback threads and will not allow new callbacks.
+    if (oboeStream->wasErrorCallbackCalled()) { // block extra error callbacks
+        LOGE("%s() multiple error callbacks called!", __func__);
+    } else if (stream != oboeStream->getUnderlyingStream()) {
+        LOGW("%s() stream already closed or closing", __func__); // might happen if there are bugs
+    } else if (sharedStream) {
+        // Handle error on a separate thread using shared pointer.
+        std::thread t(oboe_aaudio_error_thread_proc_shared, sharedStream, oboeResult);
+        t.detach();
+    } else {
+        // Handle error on a separate thread.
+        std::thread t(oboe_aaudio_error_thread_proc, oboeStream, oboeResult);
+        t.detach();
+    }
+}
+
+void AudioStreamAAudio::logUnsupportedAttributes() {
+    int sdkVersion = getSdkVersion();
+
+    // These attributes are not supported pre Android "P"
+    if (sdkVersion < __ANDROID_API_P__) {
+        if (mUsage != Usage::Media) {
+            LOGW("Usage [AudioStreamBuilder::setUsage()] "
+                 "is not supported on AAudio streams running on pre-Android P versions.");
+        }
+
+        if (mContentType != ContentType::Music) {
+            LOGW("ContentType [AudioStreamBuilder::setContentType()] "
+                 "is not supported on AAudio streams running on pre-Android P versions.");
+        }
+
+        if (mSessionId != SessionId::None) {
+            LOGW("SessionId [AudioStreamBuilder::setSessionId()] "
+                 "is not supported on AAudio streams running on pre-Android P versions.");
+        }
+    }
+}
+
+Result AudioStreamAAudio::open() {
+    Result result = Result::OK;
+
+    if (mAAudioStream != nullptr) {
+        return Result::ErrorInvalidState;
+    }
+
+    result = AudioStream::open();
+    if (result != Result::OK) {
+        return result;
+    }
+
+    AAudioStreamBuilder *aaudioBuilder;
+    result = static_cast<Result>(mLibLoader->createStreamBuilder(&aaudioBuilder));
+    if (result != Result::OK) {
+        return result;
+    }
+
+    // Do not set INPUT capacity below 4096 because that prevents us from getting a FAST track
+    // when using the Legacy data path.
+    // If the app requests > 4096 then we allow it but we are less likely to get LowLatency.
+    // See internal bug b/80308183 for more details.
+    // Fixed in Q but let's still clip the capacity because high input capacity
+    // does not increase latency.
+    int32_t capacity = mBufferCapacityInFrames;
+    constexpr int kCapacityRequiredForFastLegacyTrack = 4096; // matches value in AudioFinger
+    if (OboeGlobals::areWorkaroundsEnabled()
+            && mDirection == oboe::Direction::Input
+            && capacity != oboe::Unspecified
+            && capacity < kCapacityRequiredForFastLegacyTrack
+            && mPerformanceMode == oboe::PerformanceMode::LowLatency) {
+        capacity = kCapacityRequiredForFastLegacyTrack;
+        LOGD("AudioStreamAAudio.open() capacity changed from %d to %d for lower latency",
+             static_cast<int>(mBufferCapacityInFrames), capacity);
+    }
+    mLibLoader->builder_setBufferCapacityInFrames(aaudioBuilder, capacity);
+
+    // Channel mask was added in SC_V2. Given the corresponding channel count of selected channel
+    // mask may be different from selected channel count, the last set value will be respected.
+    // If channel count is set after channel mask, the previously set channel mask will be cleared.
+    // If channel mask is set after channel count, the channel count will be automatically
+    // calculated from selected channel mask. In that case, only set channel mask when the API
+    // is available and the channel mask is specified.
+    if (mLibLoader->builder_setChannelMask != nullptr && mChannelMask != ChannelMask::Unspecified) {
+        mLibLoader->builder_setChannelMask(aaudioBuilder,
+                                           static_cast<aaudio_channel_mask_t>(mChannelMask));
+    } else {
+        mLibLoader->builder_setChannelCount(aaudioBuilder, mChannelCount);
+    }
+    mLibLoader->builder_setDeviceId(aaudioBuilder, mDeviceId);
+    mLibLoader->builder_setDirection(aaudioBuilder, static_cast<aaudio_direction_t>(mDirection));
+    mLibLoader->builder_setFormat(aaudioBuilder, static_cast<aaudio_format_t>(mFormat));
+    mLibLoader->builder_setSampleRate(aaudioBuilder, mSampleRate);
+    mLibLoader->builder_setSharingMode(aaudioBuilder,
+                                       static_cast<aaudio_sharing_mode_t>(mSharingMode));
+    mLibLoader->builder_setPerformanceMode(aaudioBuilder,
+                                           static_cast<aaudio_performance_mode_t>(mPerformanceMode));
+
+    // These were added in P so we have to check for the function pointer.
+    if (mLibLoader->builder_setUsage != nullptr) {
+        mLibLoader->builder_setUsage(aaudioBuilder,
+                                     static_cast<aaudio_usage_t>(mUsage));
+    }
+
+    if (mLibLoader->builder_setContentType != nullptr) {
+        mLibLoader->builder_setContentType(aaudioBuilder,
+                                           static_cast<aaudio_content_type_t>(mContentType));
+    }
+
+    if (mLibLoader->builder_setInputPreset != nullptr) {
+        aaudio_input_preset_t inputPreset = mInputPreset;
+        if (getSdkVersion() <= __ANDROID_API_P__ && inputPreset == InputPreset::VoicePerformance) {
+            LOGD("InputPreset::VoicePerformance not supported before Q. Using VoiceRecognition.");
+            inputPreset = InputPreset::VoiceRecognition; // most similar preset
+        }
+        mLibLoader->builder_setInputPreset(aaudioBuilder,
+                                           static_cast<aaudio_input_preset_t>(inputPreset));
+    }
+
+    if (mLibLoader->builder_setSessionId != nullptr) {
+        mLibLoader->builder_setSessionId(aaudioBuilder,
+                                         static_cast<aaudio_session_id_t>(mSessionId));
+    }
+
+    // These were added in S so we have to check for the function pointer.
+    if (mLibLoader->builder_setPackageName != nullptr && !mPackageName.empty()) {
+        mLibLoader->builder_setPackageName(aaudioBuilder,
+                                           mPackageName.c_str());
+    }
+
+    if (mLibLoader->builder_setAttributionTag != nullptr && !mAttributionTag.empty()) {
+        mLibLoader->builder_setAttributionTag(aaudioBuilder,
+                                           mAttributionTag.c_str());
+    }
+
+    if (isDataCallbackSpecified()) {
+        mLibLoader->builder_setDataCallback(aaudioBuilder, oboe_aaudio_data_callback_proc, this);
+        mLibLoader->builder_setFramesPerDataCallback(aaudioBuilder, getFramesPerDataCallback());
+
+        if (!isErrorCallbackSpecified()) {
+            // The app did not specify a callback so we should specify
+            // our own so the stream gets closed and stopped.
+            mErrorCallback = &mDefaultErrorCallback;
+        }
+        mLibLoader->builder_setErrorCallback(aaudioBuilder, internalErrorCallback, this);
+    }
+    // Else if the data callback is not being used then the write method will return an error
+    // and the app can stop and close the stream.
+
+    // ============= OPEN THE STREAM ================
+    {
+        AAudioStream *stream = nullptr;
+        result = static_cast<Result>(mLibLoader->builder_openStream(aaudioBuilder, &stream));
+        mAAudioStream.store(stream);
+    }
+    if (result != Result::OK) {
+        // Warn developer because ErrorInternal is not very informative.
+        if (result == Result::ErrorInternal && mDirection == Direction::Input) {
+            LOGW("AudioStreamAAudio.open() may have failed due to lack of "
+                 "audio recording permission.");
+        }
+        goto error2;
+    }
+
+    // Query and cache the stream properties
+    mDeviceId = mLibLoader->stream_getDeviceId(mAAudioStream);
+    mChannelCount = mLibLoader->stream_getChannelCount(mAAudioStream);
+    mSampleRate = mLibLoader->stream_getSampleRate(mAAudioStream);
+    mFormat = static_cast<AudioFormat>(mLibLoader->stream_getFormat(mAAudioStream));
+    mSharingMode = static_cast<SharingMode>(mLibLoader->stream_getSharingMode(mAAudioStream));
+    mPerformanceMode = static_cast<PerformanceMode>(
+            mLibLoader->stream_getPerformanceMode(mAAudioStream));
+    mBufferCapacityInFrames = mLibLoader->stream_getBufferCapacity(mAAudioStream);
+    mBufferSizeInFrames = mLibLoader->stream_getBufferSize(mAAudioStream);
+    mFramesPerBurst = mLibLoader->stream_getFramesPerBurst(mAAudioStream);
+
+    // These were added in P so we have to check for the function pointer.
+    if (mLibLoader->stream_getUsage != nullptr) {
+        mUsage = static_cast<Usage>(mLibLoader->stream_getUsage(mAAudioStream));
+    }
+    if (mLibLoader->stream_getContentType != nullptr) {
+        mContentType = static_cast<ContentType>(mLibLoader->stream_getContentType(mAAudioStream));
+    }
+    if (mLibLoader->stream_getInputPreset != nullptr) {
+        mInputPreset = static_cast<InputPreset>(mLibLoader->stream_getInputPreset(mAAudioStream));
+    }
+    if (mLibLoader->stream_getSessionId != nullptr) {
+        mSessionId = static_cast<SessionId>(mLibLoader->stream_getSessionId(mAAudioStream));
+    } else {
+        mSessionId = SessionId::None;
+    }
+
+    if (mLibLoader->stream_getChannelMask != nullptr) {
+        mChannelMask = static_cast<ChannelMask>(mLibLoader->stream_getChannelMask(mAAudioStream));
+    }
+
+    LOGD("AudioStreamAAudio.open() format=%d, sampleRate=%d, capacity = %d",
+            static_cast<int>(mFormat), static_cast<int>(mSampleRate),
+            static_cast<int>(mBufferCapacityInFrames));
+
+    calculateDefaultDelayBeforeCloseMillis();
+
+error2:
+    mLibLoader->builder_delete(aaudioBuilder);
+    LOGD("AudioStreamAAudio.open: AAudioStream_Open() returned %s",
+         mLibLoader->convertResultToText(static_cast<aaudio_result_t>(result)));
+    return result;
+}
+
+Result AudioStreamAAudio::close() {
+    // Prevent two threads from closing the stream at the same time and crashing.
+    // This could occur, for example, if an application called close() at the same
+    // time that an onError callback was being executed because of a disconnect.
+    std::lock_guard<std::mutex> lock(mLock);
+
+    AudioStream::close();
+
+    AAudioStream *stream = nullptr;
+    {
+        // Wait for any methods using mAAudioStream to finish.
+        std::unique_lock<std::shared_mutex> lock2(mAAudioStreamLock);
+        // Closing will delete *mAAudioStream so we need to null out the pointer atomically.
+        stream = mAAudioStream.exchange(nullptr);
+    }
+    if (stream != nullptr) {
+        if (OboeGlobals::areWorkaroundsEnabled()) {
+            // Make sure we are really stopped. Do it under mLock
+            // so another thread cannot call requestStart() right before the close.
+            requestStop_l(stream);
+            sleepBeforeClose();
+        }
+        return static_cast<Result>(mLibLoader->stream_close(stream));
+    } else {
+        return Result::ErrorClosed;
+    }
+}
+
+static void oboe_stop_thread_proc(AudioStream *oboeStream) {
+    if (oboeStream != nullptr) {
+        oboeStream->requestStop();
+    }
+}
+
+void AudioStreamAAudio::launchStopThread() {
+    // Prevent multiple stop threads from being launched.
+    if (mStopThreadAllowed.exchange(false)) {
+        // Stop this stream on a separate thread
+        std::thread t(oboe_stop_thread_proc, this);
+        t.detach();
+    }
+}
+
+DataCallbackResult AudioStreamAAudio::callOnAudioReady(AAudioStream * /*stream*/,
+                                                                 void *audioData,
+                                                                 int32_t numFrames) {
+    DataCallbackResult result = fireDataCallback(audioData, numFrames);
+    if (result == DataCallbackResult::Continue) {
+        return result;
+    } else {
+        if (result == DataCallbackResult::Stop) {
+            LOGD("Oboe callback returned DataCallbackResult::Stop");
+        } else {
+            LOGE("Oboe callback returned unexpected value = %d", result);
+        }
+
+        // Returning Stop caused various problems before S. See #1230
+        if (OboeGlobals::areWorkaroundsEnabled() && getSdkVersion() <= __ANDROID_API_R__) {
+            launchStopThread();
+            return DataCallbackResult::Continue;
+        } else {
+            return DataCallbackResult::Stop; // OK >= API_S
+        }
+    }
+}
+
+Result AudioStreamAAudio::requestStart() {
+    std::lock_guard<std::mutex> lock(mLock);
+    AAudioStream *stream = mAAudioStream.load();
+    if (stream != nullptr) {
+        // Avoid state machine errors in O_MR1.
+        if (getSdkVersion() <= __ANDROID_API_O_MR1__) {
+            StreamState state = static_cast<StreamState>(mLibLoader->stream_getState(stream));
+            if (state == StreamState::Starting || state == StreamState::Started) {
+                // WARNING: On P, AAudio is returning ErrorInvalidState for Output and OK for Input.
+                return Result::OK;
+            }
+        }
+        if (isDataCallbackSpecified()) {
+            setDataCallbackEnabled(true);
+        }
+        mStopThreadAllowed = true;
+        return static_cast<Result>(mLibLoader->stream_requestStart(stream));
+    } else {
+        return Result::ErrorClosed;
+    }
+}
+
+Result AudioStreamAAudio::requestPause() {
+    std::lock_guard<std::mutex> lock(mLock);
+    AAudioStream *stream = mAAudioStream.load();
+    if (stream != nullptr) {
+        // Avoid state machine errors in O_MR1.
+        if (getSdkVersion() <= __ANDROID_API_O_MR1__) {
+            StreamState state = static_cast<StreamState>(mLibLoader->stream_getState(stream));
+            if (state == StreamState::Pausing || state == StreamState::Paused) {
+                return Result::OK;
+            }
+        }
+        return static_cast<Result>(mLibLoader->stream_requestPause(stream));
+    } else {
+        return Result::ErrorClosed;
+    }
+}
+
+Result AudioStreamAAudio::requestFlush() {
+    std::lock_guard<std::mutex> lock(mLock);
+    AAudioStream *stream = mAAudioStream.load();
+    if (stream != nullptr) {
+        // Avoid state machine errors in O_MR1.
+        if (getSdkVersion() <= __ANDROID_API_O_MR1__) {
+            StreamState state = static_cast<StreamState>(mLibLoader->stream_getState(stream));
+            if (state == StreamState::Flushing || state == StreamState::Flushed) {
+                return Result::OK;
+            }
+        }
+        return static_cast<Result>(mLibLoader->stream_requestFlush(stream));
+    } else {
+        return Result::ErrorClosed;
+    }
+}
+
+Result AudioStreamAAudio::requestStop() {
+    std::lock_guard<std::mutex> lock(mLock);
+    AAudioStream *stream = mAAudioStream.load();
+    if (stream != nullptr) {
+        return requestStop_l(stream);
+    } else {
+        return Result::ErrorClosed;
+    }
+}
+
+// Call under mLock
+Result AudioStreamAAudio::requestStop_l(AAudioStream *stream) {
+    // Avoid state machine errors in O_MR1.
+    if (getSdkVersion() <= __ANDROID_API_O_MR1__) {
+        StreamState state = static_cast<StreamState>(mLibLoader->stream_getState(stream));
+        if (state == StreamState::Stopping || state == StreamState::Stopped) {
+            return Result::OK;
+        }
+    }
+    return static_cast<Result>(mLibLoader->stream_requestStop(stream));
+}
+
+ResultWithValue<int32_t>   AudioStreamAAudio::write(const void *buffer,
+                                     int32_t numFrames,
+                                     int64_t timeoutNanoseconds) {
+    std::shared_lock<std::shared_mutex> lock(mAAudioStreamLock);
+    AAudioStream *stream = mAAudioStream.load();
+    if (stream != nullptr) {
+        int32_t result = mLibLoader->stream_write(mAAudioStream, buffer,
+                                                  numFrames, timeoutNanoseconds);
+        return ResultWithValue<int32_t>::createBasedOnSign(result);
+    } else {
+        return ResultWithValue<int32_t>(Result::ErrorClosed);
+    }
+}
+
+ResultWithValue<int32_t>   AudioStreamAAudio::read(void *buffer,
+                                 int32_t numFrames,
+                                 int64_t timeoutNanoseconds) {
+    std::shared_lock<std::shared_mutex> lock(mAAudioStreamLock);
+    AAudioStream *stream = mAAudioStream.load();
+    if (stream != nullptr) {
+        int32_t result = mLibLoader->stream_read(mAAudioStream, buffer,
+                                                 numFrames, timeoutNanoseconds);
+        return ResultWithValue<int32_t>::createBasedOnSign(result);
+    } else {
+        return ResultWithValue<int32_t>(Result::ErrorClosed);
+    }
+}
+
+
+// AAudioStream_waitForStateChange() can crash if it is waiting on a stream and that stream
+// is closed from another thread.  We do not want to lock the stream for the duration of the call.
+// So we call AAudioStream_waitForStateChange() with a timeout of zero so that it will not block.
+// Then we can do our own sleep with the lock unlocked.
+Result AudioStreamAAudio::waitForStateChange(StreamState currentState,
+                                        StreamState *nextState,
+                                        int64_t timeoutNanoseconds) {
+    Result oboeResult = Result::ErrorTimeout;
+    int64_t sleepTimeNanos = 20 * kNanosPerMillisecond; // arbitrary
+    aaudio_stream_state_t currentAAudioState = static_cast<aaudio_stream_state_t>(currentState);
+
+    aaudio_result_t result = AAUDIO_OK;
+    int64_t timeLeftNanos = timeoutNanoseconds;
+
+    mLock.lock();
+    while (true) {
+        // Do we still have an AAudio stream? If not then stream must have been closed.
+        AAudioStream *stream = mAAudioStream.load();
+        if (stream == nullptr) {
+            if (nextState != nullptr) {
+                *nextState = StreamState::Closed;
+            }
+            oboeResult = Result::ErrorClosed;
+            break;
+        }
+
+        // Update and query state change with no blocking.
+        aaudio_stream_state_t aaudioNextState;
+        result = mLibLoader->stream_waitForStateChange(
+                mAAudioStream,
+                currentAAudioState,
+                &aaudioNextState,
+                0); // timeout=0 for non-blocking
+        // AAudio will return AAUDIO_ERROR_TIMEOUT if timeout=0 and the state does not change.
+        if (result != AAUDIO_OK && result != AAUDIO_ERROR_TIMEOUT) {
+            oboeResult = static_cast<Result>(result);
+            break;
+        }
+#if OBOE_FIX_FORCE_STARTING_TO_STARTED
+        if (OboeGlobals::areWorkaroundsEnabled()
+            && aaudioNextState == static_cast<aaudio_stream_state_t >(StreamState::Starting)) {
+            aaudioNextState = static_cast<aaudio_stream_state_t >(StreamState::Started);
+        }
+#endif // OBOE_FIX_FORCE_STARTING_TO_STARTED
+        if (nextState != nullptr) {
+            *nextState = static_cast<StreamState>(aaudioNextState);
+        }
+        if (currentAAudioState != aaudioNextState) { // state changed?
+            oboeResult = Result::OK;
+            break;
+        }
+
+        // Did we timeout or did user ask for non-blocking?
+        if (timeLeftNanos <= 0) {
+            break;
+        }
+
+        // No change yet so sleep.
+        mLock.unlock(); // Don't sleep while locked.
+        if (sleepTimeNanos > timeLeftNanos) {
+            sleepTimeNanos = timeLeftNanos; // last little bit
+        }
+        AudioClock::sleepForNanos(sleepTimeNanos);
+        timeLeftNanos -= sleepTimeNanos;
+        mLock.lock();
+    }
+
+    mLock.unlock();
+    return oboeResult;
+}
+
+ResultWithValue<int32_t> AudioStreamAAudio::setBufferSizeInFrames(int32_t requestedFrames) {
+    int32_t adjustedFrames = requestedFrames;
+    if (adjustedFrames > mBufferCapacityInFrames) {
+        adjustedFrames = mBufferCapacityInFrames;
+    }
+    // This calls getBufferSize() so avoid recursive lock.
+    adjustedFrames = QuirksManager::getInstance().clipBufferSize(*this, adjustedFrames);
+
+    std::shared_lock<std::shared_mutex> lock(mAAudioStreamLock);
+    AAudioStream *stream = mAAudioStream.load();
+    if (stream != nullptr) {
+        int32_t newBufferSize = mLibLoader->stream_setBufferSize(mAAudioStream, adjustedFrames);
+        // Cache the result if it's valid
+        if (newBufferSize > 0) mBufferSizeInFrames = newBufferSize;
+        return ResultWithValue<int32_t>::createBasedOnSign(newBufferSize);
+    } else {
+        return ResultWithValue<int32_t>(Result::ErrorClosed);
+    }
+}
+
+StreamState AudioStreamAAudio::getState() {
+    std::shared_lock<std::shared_mutex> lock(mAAudioStreamLock);
+    AAudioStream *stream = mAAudioStream.load();
+    if (stream != nullptr) {
+        aaudio_stream_state_t aaudioState = mLibLoader->stream_getState(stream);
+#if OBOE_FIX_FORCE_STARTING_TO_STARTED
+        if (OboeGlobals::areWorkaroundsEnabled()
+            && aaudioState == AAUDIO_STREAM_STATE_STARTING) {
+            aaudioState = AAUDIO_STREAM_STATE_STARTED;
+        }
+#endif // OBOE_FIX_FORCE_STARTING_TO_STARTED
+        return static_cast<StreamState>(aaudioState);
+    } else {
+        return StreamState::Closed;
+    }
+}
+
+int32_t AudioStreamAAudio::getBufferSizeInFrames() {
+    std::shared_lock<std::shared_mutex> lock(mAAudioStreamLock);
+    AAudioStream *stream = mAAudioStream.load();
+    if (stream != nullptr) {
+        mBufferSizeInFrames = mLibLoader->stream_getBufferSize(stream);
+    }
+    return mBufferSizeInFrames;
+}
+
+void AudioStreamAAudio::updateFramesRead() {
+    std::shared_lock<std::shared_mutex> lock(mAAudioStreamLock);
+    AAudioStream *stream = mAAudioStream.load();
+// Set to 1 for debugging race condition #1180 with mAAudioStream.
+// See also DEBUG_CLOSE_RACE in OboeTester.
+// This was left in the code so that we could test the fix again easily in the future.
+// We could not trigger the race condition without adding these get calls and the sleeps.
+#define DEBUG_CLOSE_RACE 0
+#if DEBUG_CLOSE_RACE
+    // This is used when testing race conditions with close().
+    // See DEBUG_CLOSE_RACE in OboeTester
+    AudioClock::sleepForNanos(400 * kNanosPerMillisecond);
+#endif // DEBUG_CLOSE_RACE
+    if (stream != nullptr) {
+        mFramesRead = mLibLoader->stream_getFramesRead(stream);
+    }
+}
+
+void AudioStreamAAudio::updateFramesWritten() {
+    std::shared_lock<std::shared_mutex> lock(mAAudioStreamLock);
+    AAudioStream *stream = mAAudioStream.load();
+    if (stream != nullptr) {
+        mFramesWritten = mLibLoader->stream_getFramesWritten(stream);
+    }
+}
+
+ResultWithValue<int32_t> AudioStreamAAudio::getXRunCount() {
+    std::shared_lock<std::shared_mutex> lock(mAAudioStreamLock);
+    AAudioStream *stream = mAAudioStream.load();
+    if (stream != nullptr) {
+        return ResultWithValue<int32_t>::createBasedOnSign(mLibLoader->stream_getXRunCount(stream));
+    } else {
+        return ResultWithValue<int32_t>(Result::ErrorNull);
+    }
+}
+
+Result AudioStreamAAudio::getTimestamp(clockid_t clockId,
+                                   int64_t *framePosition,
+                                   int64_t *timeNanoseconds) {
+    if (getState() != StreamState::Started) {
+        return Result::ErrorInvalidState;
+    }
+    std::shared_lock<std::shared_mutex> lock(mAAudioStreamLock);
+    AAudioStream *stream = mAAudioStream.load();
+    if (stream != nullptr) {
+        return static_cast<Result>(mLibLoader->stream_getTimestamp(stream, clockId,
+                                               framePosition, timeNanoseconds));
+    } else {
+        return Result::ErrorNull;
+    }
+}
+
+ResultWithValue<double> AudioStreamAAudio::calculateLatencyMillis() {
+    // Get the time that a known audio frame was presented.
+    int64_t hardwareFrameIndex;
+    int64_t hardwareFrameHardwareTime;
+    auto result = getTimestamp(CLOCK_MONOTONIC,
+                               &hardwareFrameIndex,
+                               &hardwareFrameHardwareTime);
+    if (result != oboe::Result::OK) {
+        return ResultWithValue<double>(static_cast<Result>(result));
+    }
+
+    // Get counter closest to the app.
+    bool isOutput = (getDirection() == oboe::Direction::Output);
+    int64_t appFrameIndex = isOutput ? getFramesWritten() : getFramesRead();
+
+    // Assume that the next frame will be processed at the current time
+    using namespace std::chrono;
+    int64_t appFrameAppTime =
+            duration_cast<nanoseconds>(steady_clock::now().time_since_epoch()).count();
+
+    // Calculate the number of frames between app and hardware
+    int64_t frameIndexDelta = appFrameIndex - hardwareFrameIndex;
+
+    // Calculate the time which the next frame will be or was presented
+    int64_t frameTimeDelta = (frameIndexDelta * oboe::kNanosPerSecond) / getSampleRate();
+    int64_t appFrameHardwareTime = hardwareFrameHardwareTime + frameTimeDelta;
+
+    // The current latency is the difference in time between when the current frame is at
+    // the app and when it is at the hardware.
+    double latencyNanos = static_cast<double>(isOutput
+                          ? (appFrameHardwareTime - appFrameAppTime) // hardware is later
+                          : (appFrameAppTime - appFrameHardwareTime)); // hardware is earlier
+    double latencyMillis = latencyNanos / kNanosPerMillisecond;
+
+    return ResultWithValue<double>(latencyMillis);
+}
+
+bool AudioStreamAAudio::isMMapUsed() {
+    std::shared_lock<std::shared_mutex> lock(mAAudioStreamLock);
+    AAudioStream *stream = mAAudioStream.load();
+    if (stream != nullptr) {
+        return AAudioExtensions::getInstance().isMMapUsed(stream);
+    } else {
+        return false;
+    }
+}
+
+} // namespace oboe

vendor/github.com/ebitengine/oto/v3/internal/oboe/oboe_aaudio_AudioStreamAAudio_android.h

@@ -0,0 +1,142 @@
+/*
+ * Copyright 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef OBOE_STREAM_AAUDIO_H_
+#define OBOE_STREAM_AAUDIO_H_
+
+#include <atomic>
+#include <shared_mutex>
+#include <mutex>
+#include <thread>
+
+#include "oboe_oboe_AudioStreamBuilder_android.h"
+#include "oboe_oboe_AudioStream_android.h"
+#include "oboe_oboe_Definitions_android.h"
+#include "oboe_aaudio_AAudioLoader_android.h"
+
+namespace oboe {
+
+/**
+ * Implementation of OboeStream that uses AAudio.
+ *
+ * Do not create this class directly.
+ * Use an OboeStreamBuilder to create one.
+ */
+class AudioStreamAAudio : public AudioStream {
+public:
+    AudioStreamAAudio();
+    explicit AudioStreamAAudio(const AudioStreamBuilder &builder);
+
+    virtual ~AudioStreamAAudio() = default;
+
+    /**
+     *
+     * @return true if AAudio is supported on this device.
+     */
+    static bool isSupported();
+
+    // These functions override methods in AudioStream.
+    // See AudioStream for documentation.
+    Result open() override;
+    Result close() override;
+
+    Result requestStart() override;
+    Result requestPause() override;
+    Result requestFlush() override;
+    Result requestStop() override;
+
+    ResultWithValue<int32_t> write(const void *buffer,
+                  int32_t numFrames,
+                  int64_t timeoutNanoseconds) override;
+
+    ResultWithValue<int32_t> read(void *buffer,
+                 int32_t numFrames,
+                 int64_t timeoutNanoseconds) override;
+
+    ResultWithValue<int32_t> setBufferSizeInFrames(int32_t requestedFrames) override;
+    int32_t getBufferSizeInFrames() override;
+    ResultWithValue<int32_t> getXRunCount()  override;
+    bool isXRunCountSupported() const override { return true; }
+
+    ResultWithValue<double> calculateLatencyMillis() override;
+
+    Result waitForStateChange(StreamState currentState,
+                              StreamState *nextState,
+                              int64_t timeoutNanoseconds) override;
+
+    Result getTimestamp(clockid_t clockId,
+                                       int64_t *framePosition,
+                                       int64_t *timeNanoseconds) override;
+
+    StreamState getState() override;
+
+    AudioApi getAudioApi() const override {
+        return AudioApi::AAudio;
+    }
+
+    DataCallbackResult callOnAudioReady(AAudioStream *stream,
+                                                   void *audioData,
+                                                   int32_t numFrames);
+
+    bool isMMapUsed();
+
+protected:
+    static void internalErrorCallback(
+            AAudioStream *stream,
+            void *userData,
+            aaudio_result_t error);
+
+    void *getUnderlyingStream() const override {
+        return mAAudioStream.load();
+    }
+
+    void updateFramesRead() override;
+    void updateFramesWritten() override;
+
+    void logUnsupportedAttributes();
+
+private:
+    // Must call under mLock. And stream must NOT be nullptr.
+    Result requestStop_l(AAudioStream *stream);
+
+    /**
+     * Launch a thread that will stop the stream.
+     */
+    void launchStopThread();
+
+public:
+    int32_t getMDelayBeforeCloseMillis() const;
+
+    void setDelayBeforeCloseMillis(int32_t mDelayBeforeCloseMillis);
+
+private:
+
+    std::atomic<bool>    mCallbackThreadEnabled;
+    std::atomic<bool>    mStopThreadAllowed{false};
+
+    // pointer to the underlying 'C' AAudio stream, valid if open, null if closed
+    std::atomic<AAudioStream *> mAAudioStream{nullptr};
+    std::shared_mutex           mAAudioStreamLock; // to protect mAAudioStream while closing
+
+    static AAudioLoader *mLibLoader;
+
+    // We may not use this but it is so small that it is not worth allocating dynamically.
+    AudioStreamErrorCallback mDefaultErrorCallback;
+};
+
+} // namespace oboe
+
+#endif // OBOE_STREAM_AAUDIO_H_

vendor/github.com/ebitengine/oto/v3/internal/oboe/oboe_common_AudioClock_android.h

@@ -0,0 +1,75 @@
+/*
+ * Copyright (C) 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef OBOE_AUDIO_CLOCK_H
+#define OBOE_AUDIO_CLOCK_H
+
+#include <sys/types.h>
+#include <ctime>
+#include "oboe_oboe_Definitions_android.h"
+
+namespace oboe {
+
+// TODO: Move this class into the public headers because it is useful when calculating stream latency
+class AudioClock {
+public:
+    static int64_t getNanoseconds(clockid_t clockId = CLOCK_MONOTONIC) {
+        struct timespec time;
+        int result = clock_gettime(clockId, &time);
+        if (result < 0) {
+            return result;
+        }
+        return (time.tv_sec * kNanosPerSecond) + time.tv_nsec;
+    }
+
+    /**
+     * Sleep until the specified time.
+     *
+     * @param nanoTime time to wake up
+     * @param clockId CLOCK_MONOTONIC is default
+     * @return 0 or a negative error, eg. -EINTR
+     */
+
+    static int sleepUntilNanoTime(int64_t nanoTime, clockid_t clockId = CLOCK_MONOTONIC) {
+        struct timespec time;
+        time.tv_sec = nanoTime / kNanosPerSecond;
+        time.tv_nsec = nanoTime - (time.tv_sec * kNanosPerSecond);
+        return 0 - clock_nanosleep(clockId, TIMER_ABSTIME, &time, NULL);
+    }
+
+    /**
+     * Sleep for the specified number of nanoseconds in real-time.
+     * Return immediately with 0 if a negative nanoseconds is specified.
+     *
+     * @param nanoseconds time to sleep
+     * @param clockId CLOCK_REALTIME is default
+     * @return 0 or a negative error, eg. -EINTR
+     */
+
+    static int sleepForNanos(int64_t nanoseconds, clockid_t clockId = CLOCK_REALTIME) {
+        if (nanoseconds > 0) {
+            struct timespec time;
+            time.tv_sec = nanoseconds / kNanosPerSecond;
+            time.tv_nsec = nanoseconds - (time.tv_sec * kNanosPerSecond);
+            return 0 - clock_nanosleep(clockId, 0, &time, NULL);
+        }
+        return 0;
+    }
+};
+
+} // namespace oboe
+
+#endif //OBOE_AUDIO_CLOCK_H

vendor/github.com/ebitengine/oto/v3/internal/oboe/oboe_common_AudioSourceCaller_android.cpp

@@ -0,0 +1,38 @@
+/*
+ * Copyright 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include "oboe_common_AudioSourceCaller_android.h"
+
+using namespace oboe;
+using namespace flowgraph;
+
+int32_t AudioSourceCaller::onProcessFixedBlock(uint8_t *buffer, int32_t numBytes) {
+    AudioStreamDataCallback *callback = mStream->getDataCallback();
+    int32_t result = 0;
+    int32_t numFrames = numBytes / mStream->getBytesPerFrame();
+    if (callback != nullptr) {
+        DataCallbackResult callbackResult = callback->onAudioReady(mStream, buffer, numFrames);
+        // onAudioReady() does not return the number of bytes processed so we have to assume all.
+        result = (callbackResult == DataCallbackResult::Continue)
+                ? numBytes
+                : -1;
+    } else {
+        auto readResult = mStream->read(buffer, numFrames, mTimeoutNanos);
+        if (!readResult) return (int32_t) readResult.error();
+        result = readResult.value() * mStream->getBytesPerFrame();
+    }
+    return result;
+}

vendor/github.com/ebitengine/oto/v3/internal/oboe/oboe_common_AudioSourceCaller_android.h

@@ -0,0 +1,83 @@
+/*
+ * Copyright 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef OBOE_AUDIO_SOURCE_CALLER_H
+#define OBOE_AUDIO_SOURCE_CALLER_H
+
+#include "oboe_common_OboeDebug_android.h"
+#include "oboe_oboe_Oboe_android.h"
+
+#include "oboe_flowgraph_FlowGraphNode_android.h"
+#include "oboe_common_FixedBlockReader_android.h"
+
+namespace oboe {
+
+class AudioStreamCallback;
+class AudioStream;
+
+/**
+ * For output streams that use a callback, call the application for more data.
+ * For input streams that do not use a callback, read from the stream.
+ */
+class AudioSourceCaller : public flowgraph::FlowGraphSource, public FixedBlockProcessor {
+public:
+    AudioSourceCaller(int32_t channelCount, int32_t framesPerCallback, int32_t bytesPerSample)
+            : FlowGraphSource(channelCount)
+            , mBlockReader(*this) {
+        mBlockReader.open(channelCount * framesPerCallback * bytesPerSample);
+    }
+
+    /**
+     * Set the stream to use as a source of data.
+     * @param stream
+     */
+    void setStream(oboe::AudioStream *stream) {
+        mStream = stream;
+    }
+
+    oboe::AudioStream *getStream() {
+        return mStream;
+    }
+
+    /**
+     * Timeout value to use when calling audioStream->read().
+     * @param timeoutNanos Zero for no timeout or time in nanoseconds.
+     */
+    void setTimeoutNanos(int64_t timeoutNanos) {
+        mTimeoutNanos = timeoutNanos;
+    }
+
+    int64_t getTimeoutNanos() const {
+        return mTimeoutNanos;
+    }
+
+    /**
+     * Called internally for block size adaptation.
+     * @param buffer
+     * @param numBytes
+     * @return
+     */
+    int32_t onProcessFixedBlock(uint8_t *buffer, int32_t numBytes) override;
+
+protected:
+    oboe::AudioStream         *mStream = nullptr;
+    int64_t                    mTimeoutNanos = 0;
+
+    FixedBlockReader           mBlockReader;
+};
+
+}
+#endif //OBOE_AUDIO_SOURCE_CALLER_H

vendor/github.com/ebitengine/oto/v3/internal/oboe/oboe_common_AudioStreamBuilder_android.cpp

@@ -0,0 +1,224 @@
+/*
+ * Copyright 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <sys/types.h>
+
+
+#include "oboe_aaudio_AAudioExtensions_android.h"
+#include "oboe_aaudio_AudioStreamAAudio_android.h"
+#include "oboe_common_FilterAudioStream_android.h"
+#include "oboe_common_OboeDebug_android.h"
+#include "oboe_oboe_Oboe_android.h"
+#include "oboe_oboe_AudioStreamBuilder_android.h"
+#include "oboe_opensles_AudioInputStreamOpenSLES_android.h"
+#include "oboe_opensles_AudioOutputStreamOpenSLES_android.h"
+#include "oboe_opensles_AudioStreamOpenSLES_android.h"
+#include "oboe_common_QuirksManager_android.h"
+
+bool oboe::OboeGlobals::mWorkaroundsEnabled = true;
+
+namespace oboe {
+
+/**
+ * The following default values are used when oboe does not have any better way of determining the optimal values
+ * for an audio stream. This can happen when:
+ *
+ * - Client is creating a stream on API < 26 (OpenSLES) but has not supplied the optimal sample
+ * rate and/or frames per burst
+ * - Client is creating a stream on API 16 (OpenSLES) where AudioManager.PROPERTY_OUTPUT_* values
+ * are not available
+ */
+int32_t DefaultStreamValues::SampleRate = 48000; // Common rate for mobile audio and video
+int32_t DefaultStreamValues::FramesPerBurst = 192; // 4 msec at 48000 Hz
+int32_t DefaultStreamValues::ChannelCount = 2; // Stereo
+
+constexpr int kBufferSizeInBurstsForLowLatencyStreams = 2;
+
+#ifndef OBOE_ENABLE_AAUDIO
+// Set OBOE_ENABLE_AAUDIO to 0 if you want to disable the AAudio API.
+// This might be useful if you want to force all the unit tests to use OpenSL ES.
+#define OBOE_ENABLE_AAUDIO 1
+#endif
+
+bool AudioStreamBuilder::isAAudioSupported() {
+    return AudioStreamAAudio::isSupported() && OBOE_ENABLE_AAUDIO;
+}
+
+bool AudioStreamBuilder::isAAudioRecommended() {
+    // See https://github.com/google/oboe/issues/40,
+    // AAudio may not be stable on Android O, depending on how it is used.
+    // To be safe, use AAudio only on O_MR1 and above.
+    return (getSdkVersion() >= __ANDROID_API_O_MR1__) && isAAudioSupported();
+}
+
+AudioStream *AudioStreamBuilder::build() {
+    AudioStream *stream = nullptr;
+    if (isAAudioRecommended() && mAudioApi != AudioApi::OpenSLES) {
+        stream = new AudioStreamAAudio(*this);
+    } else if (isAAudioSupported() && mAudioApi == AudioApi::AAudio) {
+        stream = new AudioStreamAAudio(*this);
+        LOGE("Creating AAudio stream on 8.0 because it was specified. This is error prone.");
+    } else {
+        if (getDirection() == oboe::Direction::Output) {
+            stream = new AudioOutputStreamOpenSLES(*this);
+        } else if (getDirection() == oboe::Direction::Input) {
+            stream = new AudioInputStreamOpenSLES(*this);
+        }
+    }
+    return stream;
+}
+
+bool AudioStreamBuilder::isCompatible(AudioStreamBase &other) {
+    return (getSampleRate() == oboe::Unspecified || getSampleRate() == other.getSampleRate())
+           && (getFormat() == (AudioFormat)oboe::Unspecified || getFormat() == other.getFormat())
+           && (getFramesPerDataCallback() == oboe::Unspecified || getFramesPerDataCallback() == other.getFramesPerDataCallback())
+           && (getChannelCount() == oboe::Unspecified || getChannelCount() == other.getChannelCount());
+}
+
+Result AudioStreamBuilder::openStream(AudioStream **streamPP) {
+    auto result = isValidConfig();
+    if (result != Result::OK) {
+        LOGW("%s() invalid config %d", __func__, result);
+        return result;
+    }
+
+    LOGI("%s() %s -------- %s --------",
+         __func__, getDirection() == Direction::Input ? "INPUT" : "OUTPUT", getVersionText());
+
+    if (streamPP == nullptr) {
+        return Result::ErrorNull;
+    }
+    *streamPP = nullptr;
+
+    AudioStream *streamP = nullptr;
+
+    // Maybe make a FilterInputStream.
+    AudioStreamBuilder childBuilder(*this);
+    // Check need for conversion and modify childBuilder for optimal stream.
+    bool conversionNeeded = QuirksManager::getInstance().isConversionNeeded(*this, childBuilder);
+    // Do we need to make a child stream and convert.
+    if (conversionNeeded) {
+        AudioStream *tempStream;
+        result = childBuilder.openStream(&tempStream);
+        if (result != Result::OK) {
+            return result;
+        }
+
+        if (isCompatible(*tempStream)) {
+            // The child stream would work as the requested stream so we can just use it directly.
+            *streamPP = tempStream;
+            return result;
+        } else {
+            AudioStreamBuilder parentBuilder = *this;
+            // Build a stream that is as close as possible to the childStream.
+            if (getFormat() == oboe::AudioFormat::Unspecified) {
+                parentBuilder.setFormat(tempStream->getFormat());
+            }
+            if (getChannelCount() == oboe::Unspecified) {
+                parentBuilder.setChannelCount(tempStream->getChannelCount());
+            }
+            if (getSampleRate() == oboe::Unspecified) {
+                parentBuilder.setSampleRate(tempStream->getSampleRate());
+            }
+            if (getFramesPerDataCallback() == oboe::Unspecified) {
+                parentBuilder.setFramesPerCallback(tempStream->getFramesPerDataCallback());
+            }
+
+            // Use childStream in a FilterAudioStream.
+            LOGI("%s() create a FilterAudioStream for data conversion.", __func__);
+            FilterAudioStream *filterStream = new FilterAudioStream(parentBuilder, tempStream);
+            result = filterStream->configureFlowGraph();
+            if (result !=  Result::OK) {
+                filterStream->close();
+                delete filterStream;
+                // Just open streamP the old way.
+            } else {
+                streamP = static_cast<AudioStream *>(filterStream);
+            }
+        }
+    }
+
+    if (streamP == nullptr) {
+        streamP = build();
+        if (streamP == nullptr) {
+            return Result::ErrorNull;
+        }
+    }
+
+    // If MMAP has a problem in this case then disable it temporarily.
+    bool wasMMapOriginallyEnabled = AAudioExtensions::getInstance().isMMapEnabled();
+    bool wasMMapTemporarilyDisabled = false;
+    if (wasMMapOriginallyEnabled) {
+        bool isMMapSafe = QuirksManager::getInstance().isMMapSafe(childBuilder);
+        if (!isMMapSafe) {
+            AAudioExtensions::getInstance().setMMapEnabled(false);
+            wasMMapTemporarilyDisabled = true;
+        }
+    }
+    result = streamP->open();
+    if (wasMMapTemporarilyDisabled) {
+        AAudioExtensions::getInstance().setMMapEnabled(wasMMapOriginallyEnabled); // restore original
+    }
+    if (result == Result::OK) {
+
+        int32_t  optimalBufferSize = -1;
+        // Use a reasonable default buffer size.
+        if (streamP->getDirection() == Direction::Input) {
+            // For input, small size does not improve latency because the stream is usually
+            // run close to empty. And a low size can result in XRuns so always use the maximum.
+            optimalBufferSize = streamP->getBufferCapacityInFrames();
+        } else if (streamP->getPerformanceMode() == PerformanceMode::LowLatency
+                && streamP->getDirection() == Direction::Output)  { // Output check is redundant.
+            optimalBufferSize = streamP->getFramesPerBurst() *
+                                    kBufferSizeInBurstsForLowLatencyStreams;
+        }
+        if (optimalBufferSize >= 0) {
+            auto setBufferResult = streamP->setBufferSizeInFrames(optimalBufferSize);
+            if (!setBufferResult) {
+                LOGW("Failed to setBufferSizeInFrames(%d). Error was %s",
+                     optimalBufferSize,
+                     convertToText(setBufferResult.error()));
+            }
+        }
+
+        *streamPP = streamP;
+    } else {
+        delete streamP;
+    }
+    return result;
+}
+
+Result AudioStreamBuilder::openManagedStream(oboe::ManagedStream &stream) {
+    stream.reset();
+    AudioStream *streamptr;
+    auto result = openStream(&streamptr);
+    stream.reset(streamptr);
+    return result;
+}
+
+Result AudioStreamBuilder::openStream(std::shared_ptr<AudioStream> &sharedStream) {
+    sharedStream.reset();
+    AudioStream *streamptr;
+    auto result = openStream(&streamptr);
+    if (result == Result::OK) {
+        sharedStream.reset(streamptr);
+        // Save a weak_ptr in the stream for use with callbacks.
+        streamptr->setWeakThis(sharedStream);
+    }
+    return result;
+}
+
+} // namespace oboe

vendor/github.com/ebitengine/oto/v3/internal/oboe/oboe_common_AudioStream_android.cpp

@@ -0,0 +1,208 @@
+/*
+ * Copyright 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <sys/types.h>
+#include <pthread.h>
+#include <thread>
+
+#include "oboe_oboe_AudioStream_android.h"
+#include "oboe_common_OboeDebug_android.h"
+#include "oboe_common_AudioClock_android.h"
+#include "oboe_oboe_Utilities_android.h"
+
+namespace oboe {
+
+/*
+ * AudioStream
+ */
+AudioStream::AudioStream(const AudioStreamBuilder &builder)
+        : AudioStreamBase(builder) {
+}
+
+Result AudioStream::close() {
+    // Update local counters so they can be read after the close.
+    updateFramesWritten();
+    updateFramesRead();
+    return Result::OK;
+}
+
+// Call this from fireDataCallback() if you want to monitor CPU scheduler.
+void AudioStream::checkScheduler() {
+    int scheduler = sched_getscheduler(0) & ~SCHED_RESET_ON_FORK; // for current thread
+    if (scheduler != mPreviousScheduler) {
+        LOGD("AudioStream::%s() scheduler = %s", __func__,
+                ((scheduler == SCHED_FIFO) ? "SCHED_FIFO" :
+                ((scheduler == SCHED_OTHER) ? "SCHED_OTHER" :
+                ((scheduler == SCHED_RR) ? "SCHED_RR" : "UNKNOWN")))
+        );
+        mPreviousScheduler = scheduler;
+    }
+}
+
+DataCallbackResult AudioStream::fireDataCallback(void *audioData, int32_t numFrames) {
+    if (!isDataCallbackEnabled()) {
+        LOGW("AudioStream::%s() called with data callback disabled!", __func__);
+        return DataCallbackResult::Stop; // Should not be getting called
+    }
+
+    DataCallbackResult result;
+    if (mDataCallback) {
+        result = mDataCallback->onAudioReady(this, audioData, numFrames);
+    } else {
+        result = onDefaultCallback(audioData, numFrames);
+    }
+    // On Oreo, we might get called after returning stop.
+    // So block that here.
+    setDataCallbackEnabled(result == DataCallbackResult::Continue);
+
+    return result;
+}
+
+Result AudioStream::waitForStateTransition(StreamState startingState,
+                                           StreamState endingState,
+                                           int64_t timeoutNanoseconds)
+{
+    StreamState state;
+    {
+        std::lock_guard<std::mutex> lock(mLock);
+        state = getState();
+        if (state == StreamState::Closed) {
+            return Result::ErrorClosed;
+        } else if (state == StreamState::Disconnected) {
+            return Result::ErrorDisconnected;
+        }
+    }
+
+    StreamState nextState = state;
+    // TODO Should this be a while()?!
+    if (state == startingState && state != endingState) {
+        Result result = waitForStateChange(state, &nextState, timeoutNanoseconds);
+        if (result != Result::OK) {
+            return result;
+        }
+    }
+
+    if (nextState != endingState) {
+        return Result::ErrorInvalidState;
+    } else {
+        return Result::OK;
+    }
+}
+
+Result AudioStream::start(int64_t timeoutNanoseconds)
+{
+    Result result = requestStart();
+    if (result != Result::OK) return result;
+    if (timeoutNanoseconds <= 0) return result;
+    return waitForStateTransition(StreamState::Starting,
+                                  StreamState::Started, timeoutNanoseconds);
+}
+
+Result AudioStream::pause(int64_t timeoutNanoseconds)
+{
+    Result result = requestPause();
+    if (result != Result::OK) return result;
+    if (timeoutNanoseconds <= 0) return result;
+    return waitForStateTransition(StreamState::Pausing,
+                                  StreamState::Paused, timeoutNanoseconds);
+}
+
+Result AudioStream::flush(int64_t timeoutNanoseconds)
+{
+    Result result = requestFlush();
+    if (result != Result::OK) return result;
+    if (timeoutNanoseconds <= 0) return result;
+    return waitForStateTransition(StreamState::Flushing,
+                                  StreamState::Flushed, timeoutNanoseconds);
+}
+
+Result AudioStream::stop(int64_t timeoutNanoseconds)
+{
+    Result result = requestStop();
+    if (result != Result::OK) return result;
+    if (timeoutNanoseconds <= 0) return result;
+    return waitForStateTransition(StreamState::Stopping,
+                                  StreamState::Stopped, timeoutNanoseconds);
+}
+
+int32_t AudioStream::getBytesPerSample() const {
+    return convertFormatToSizeInBytes(mFormat);
+}
+
+int64_t AudioStream::getFramesRead() {
+    updateFramesRead();
+    return mFramesRead;
+}
+
+int64_t AudioStream::getFramesWritten() {
+    updateFramesWritten();
+    return mFramesWritten;
+}
+
+ResultWithValue<int32_t> AudioStream::getAvailableFrames() {
+    int64_t readCounter = getFramesRead();
+    if (readCounter < 0) return ResultWithValue<int32_t>::createBasedOnSign(readCounter);
+    int64_t writeCounter = getFramesWritten();
+    if (writeCounter < 0) return ResultWithValue<int32_t>::createBasedOnSign(writeCounter);
+    int32_t framesAvailable = writeCounter - readCounter;
+    return ResultWithValue<int32_t>(framesAvailable);
+}
+
+ResultWithValue<int32_t> AudioStream::waitForAvailableFrames(int32_t numFrames,
+        int64_t timeoutNanoseconds) {
+    if (numFrames == 0) return Result::OK;
+    if (numFrames < 0) return Result::ErrorOutOfRange;
+
+    int64_t framesAvailable = 0;
+    int64_t burstInNanos = getFramesPerBurst() * kNanosPerSecond / getSampleRate();
+    bool ready = false;
+    int64_t deadline = AudioClock::getNanoseconds() + timeoutNanoseconds;
+    do {
+        ResultWithValue<int32_t> result = getAvailableFrames();
+        if (!result) return result;
+        framesAvailable = result.value();
+        ready = (framesAvailable >= numFrames);
+        if (!ready) {
+            int64_t now = AudioClock::getNanoseconds();
+            if (now > deadline) break;
+            AudioClock::sleepForNanos(burstInNanos);
+        }
+    } while (!ready);
+    return (!ready)
+            ? ResultWithValue<int32_t>(Result::ErrorTimeout)
+            : ResultWithValue<int32_t>(framesAvailable);
+}
+
+ResultWithValue<FrameTimestamp> AudioStream::getTimestamp(clockid_t clockId) {
+    FrameTimestamp frame;
+    Result result = getTimestamp(clockId, &frame.position, &frame.timestamp);
+    if (result == Result::OK){
+        return ResultWithValue<FrameTimestamp>(frame);
+    } else {
+        return ResultWithValue<FrameTimestamp>(static_cast<Result>(result));
+    }
+}
+
+void AudioStream::calculateDefaultDelayBeforeCloseMillis() {
+    // Calculate delay time before close based on burst duration.
+    // Start with a burst duration then add 1 msec as a safety margin.
+    mDelayBeforeCloseMillis = std::max(kMinDelayBeforeCloseMillis,
+                                       1 + ((mFramesPerBurst * 1000) / getSampleRate()));
+    LOGD("calculateDefaultDelayBeforeCloseMillis() default = %d",
+         static_cast<int>(mDelayBeforeCloseMillis));
+}
+
+} // namespace oboe

vendor/github.com/ebitengine/oto/v3/internal/oboe/oboe_common_DataConversionFlowGraph_android.cpp

@@ -0,0 +1,267 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <memory>
+
+#include "oboe_common_OboeDebug_android.h"
+#include "oboe_common_DataConversionFlowGraph_android.h"
+#include "oboe_common_SourceFloatCaller_android.h"
+#include "oboe_common_SourceI16Caller_android.h"
+#include "oboe_common_SourceI24Caller_android.h"
+#include "oboe_common_SourceI32Caller_android.h"
+
+#include "oboe_flowgraph_ClipToRange_android.h"
+#include "oboe_flowgraph_MonoToMultiConverter_android.h"
+#include "oboe_flowgraph_MultiToMonoConverter_android.h"
+#include "oboe_flowgraph_RampLinear_android.h"
+#include "oboe_flowgraph_SinkFloat_android.h"
+#include "oboe_flowgraph_SinkI16_android.h"
+#include "oboe_flowgraph_SinkI24_android.h"
+#include "oboe_flowgraph_SinkI32_android.h"
+#include "oboe_flowgraph_SourceFloat_android.h"
+#include "oboe_flowgraph_SourceI16_android.h"
+#include "oboe_flowgraph_SourceI24_android.h"
+#include "oboe_flowgraph_SourceI32_android.h"
+#include "oboe_flowgraph_SampleRateConverter_android.h"
+
+using namespace oboe;
+using namespace flowgraph;
+using namespace resampler;
+
+void DataConversionFlowGraph::setSource(const void *buffer, int32_t numFrames) {
+    mSource->setData(buffer, numFrames);
+}
+
+static MultiChannelResampler::Quality convertOboeSRQualityToMCR(SampleRateConversionQuality quality) {
+    switch (quality) {
+        case SampleRateConversionQuality::Fastest:
+            return MultiChannelResampler::Quality::Fastest;
+        case SampleRateConversionQuality::Low:
+            return MultiChannelResampler::Quality::Low;
+        default:
+        case SampleRateConversionQuality::Medium:
+            return MultiChannelResampler::Quality::Medium;
+        case SampleRateConversionQuality::High:
+            return MultiChannelResampler::Quality::High;
+        case SampleRateConversionQuality::Best:
+            return MultiChannelResampler::Quality::Best;
+    }
+}
+
+// Chain together multiple processors.
+// Callback Output
+//     Use SourceCaller that calls original app callback from the flowgraph.
+//     The child callback from FilteredAudioStream read()s from the flowgraph.
+// Callback Input
+//     Child callback from FilteredAudioStream writes()s to the flowgraph.
+//     The output of the flowgraph goes through a BlockWriter to the app callback.
+// Blocking Write
+//     Write buffer is set on an AudioSource.
+//     Data is pulled through the graph and written to the child stream.
+// Blocking Read
+//     Reads in a loop from the flowgraph Sink to fill the read buffer.
+//     A SourceCaller then does a blocking read from the child Stream.
+//
+Result DataConversionFlowGraph::configure(AudioStream *sourceStream, AudioStream *sinkStream) {
+
+    FlowGraphPortFloatOutput *lastOutput = nullptr;
+
+    bool isOutput = sourceStream->getDirection() == Direction::Output;
+    bool isInput = !isOutput;
+    mFilterStream = isOutput ? sourceStream : sinkStream;
+
+    AudioFormat sourceFormat = sourceStream->getFormat();
+    int32_t sourceChannelCount = sourceStream->getChannelCount();
+    int32_t sourceSampleRate = sourceStream->getSampleRate();
+    int32_t sourceFramesPerCallback = sourceStream->getFramesPerDataCallback();
+
+    AudioFormat sinkFormat = sinkStream->getFormat();
+    int32_t sinkChannelCount = sinkStream->getChannelCount();
+    int32_t sinkSampleRate = sinkStream->getSampleRate();
+    int32_t sinkFramesPerCallback = sinkStream->getFramesPerDataCallback();
+
+    LOGI("%s() flowgraph converts channels: %d to %d, format: %d to %d"
+         ", rate: %d to %d, cbsize: %d to %d, qual = %d",
+            __func__,
+            sourceChannelCount, sinkChannelCount,
+            sourceFormat, sinkFormat,
+            sourceSampleRate, sinkSampleRate,
+            sourceFramesPerCallback, sinkFramesPerCallback,
+            sourceStream->getSampleRateConversionQuality());
+
+    // Source
+    // IF OUTPUT and using a callback then call back to the app using a SourceCaller.
+    // OR IF INPUT and NOT using a callback then read from the child stream using a SourceCaller.
+    bool isDataCallbackSpecified = sourceStream->isDataCallbackSpecified();
+    if ((isDataCallbackSpecified && isOutput)
+        || (!isDataCallbackSpecified && isInput)) {
+        int32_t actualSourceFramesPerCallback = (sourceFramesPerCallback == kUnspecified)
+                ? sourceStream->getFramesPerBurst()
+                : sourceFramesPerCallback;
+        switch (sourceFormat) {
+            case AudioFormat::Float:
+                mSourceCaller = std::make_unique<SourceFloatCaller>(sourceChannelCount,
+                                                                    actualSourceFramesPerCallback);
+                break;
+            case AudioFormat::I16:
+                mSourceCaller = std::make_unique<SourceI16Caller>(sourceChannelCount,
+                                                                  actualSourceFramesPerCallback);
+                break;
+            case AudioFormat::I24:
+                mSourceCaller = std::make_unique<SourceI24Caller>(sourceChannelCount,
+                                                                  actualSourceFramesPerCallback);
+                break;
+            case AudioFormat::I32:
+                mSourceCaller = std::make_unique<SourceI32Caller>(sourceChannelCount,
+                                                                  actualSourceFramesPerCallback);
+                break;
+            default:
+                LOGE("%s() Unsupported source caller format = %d", __func__, sourceFormat);
+                return Result::ErrorIllegalArgument;
+        }
+        mSourceCaller->setStream(sourceStream);
+        lastOutput = &mSourceCaller->output;
+    } else {
+        // IF OUTPUT and NOT using a callback then write to the child stream using a BlockWriter.
+        // OR IF INPUT and using a callback then write to the app using a BlockWriter.
+        switch (sourceFormat) {
+            case AudioFormat::Float:
+                mSource = std::make_unique<SourceFloat>(sourceChannelCount);
+                break;
+            case AudioFormat::I16:
+                mSource = std::make_unique<SourceI16>(sourceChannelCount);
+                break;
+            case AudioFormat::I24:
+                mSource = std::make_unique<SourceI24>(sourceChannelCount);
+                break;
+            case AudioFormat::I32:
+                mSource = std::make_unique<SourceI32>(sourceChannelCount);
+                break;
+            default:
+                LOGE("%s() Unsupported source format = %d", __func__, sourceFormat);
+                return Result::ErrorIllegalArgument;
+        }
+        if (isInput) {
+            int32_t actualSinkFramesPerCallback = (sinkFramesPerCallback == kUnspecified)
+                    ? sinkStream->getFramesPerBurst()
+                    : sinkFramesPerCallback;
+            // The BlockWriter is after the Sink so use the SinkStream size.
+            mBlockWriter.open(actualSinkFramesPerCallback * sinkStream->getBytesPerFrame());
+            mAppBuffer = std::make_unique<uint8_t[]>(
+                    kDefaultBufferSize * sinkStream->getBytesPerFrame());
+        }
+        lastOutput = &mSource->output;
+    }
+
+    // If we are going to reduce the number of channels then do it before the
+    // sample rate converter.
+    if (sourceChannelCount > sinkChannelCount) {
+        if (sinkChannelCount == 1) {
+            mMultiToMonoConverter = std::make_unique<MultiToMonoConverter>(sourceChannelCount);
+            lastOutput->connect(&mMultiToMonoConverter->input);
+            lastOutput = &mMultiToMonoConverter->output;
+        } else {
+            mChannelCountConverter = std::make_unique<ChannelCountConverter>(
+                    sourceChannelCount,
+                    sinkChannelCount);
+            lastOutput->connect(&mChannelCountConverter->input);
+            lastOutput = &mChannelCountConverter->output;
+        }
+    }
+
+    // Sample Rate conversion
+    if (sourceSampleRate != sinkSampleRate) {
+        // Create a resampler to do the math.
+        mResampler.reset(MultiChannelResampler::make(lastOutput->getSamplesPerFrame(),
+                                                     sourceSampleRate,
+                                                     sinkSampleRate,
+                                                     convertOboeSRQualityToMCR(
+                                                             sourceStream->getSampleRateConversionQuality())));
+        // Make a flowgraph node that uses the resampler.
+        mRateConverter = std::make_unique<SampleRateConverter>(lastOutput->getSamplesPerFrame(),
+                                                               *mResampler.get());
+        lastOutput->connect(&mRateConverter->input);
+        lastOutput = &mRateConverter->output;
+    }
+
+    // Expand the number of channels if required.
+    if (sourceChannelCount < sinkChannelCount) {
+        if (sourceChannelCount == 1) {
+            mMonoToMultiConverter = std::make_unique<MonoToMultiConverter>(sinkChannelCount);
+            lastOutput->connect(&mMonoToMultiConverter->input);
+            lastOutput = &mMonoToMultiConverter->output;
+        } else {
+            mChannelCountConverter = std::make_unique<ChannelCountConverter>(
+                    sourceChannelCount,
+                    sinkChannelCount);
+            lastOutput->connect(&mChannelCountConverter->input);
+            lastOutput = &mChannelCountConverter->output;
+        }
+    }
+
+    // Sink
+    switch (sinkFormat) {
+        case AudioFormat::Float:
+            mSink = std::make_unique<SinkFloat>(sinkChannelCount);
+            break;
+        case AudioFormat::I16:
+            mSink = std::make_unique<SinkI16>(sinkChannelCount);
+            break;
+        case AudioFormat::I24:
+            mSink = std::make_unique<SinkI24>(sinkChannelCount);
+            break;
+        case AudioFormat::I32:
+            mSink = std::make_unique<SinkI32>(sinkChannelCount);
+            break;
+        default:
+            LOGE("%s() Unsupported sink format = %d", __func__, sinkFormat);
+            return Result::ErrorIllegalArgument;;
+    }
+    lastOutput->connect(&mSink->input);
+
+    return Result::OK;
+}
+
+int32_t DataConversionFlowGraph::read(void *buffer, int32_t numFrames, int64_t timeoutNanos) {
+    if (mSourceCaller) {
+        mSourceCaller->setTimeoutNanos(timeoutNanos);
+    }
+    int32_t numRead = mSink->read(buffer, numFrames);
+    return numRead;
+}
+
+// This is similar to pushing data through the flowgraph.
+int32_t DataConversionFlowGraph::write(void *inputBuffer, int32_t numFrames) {
+    // Put the data from the input at the head of the flowgraph.
+    mSource->setData(inputBuffer, numFrames);
+    while (true) {
+        // Pull and read some data in app format into a small buffer.
+        int32_t framesRead = mSink->read(mAppBuffer.get(), flowgraph::kDefaultBufferSize);
+        if (framesRead <= 0) break;
+        // Write to a block adapter, which will call the destination whenever it has enough data.
+        int32_t bytesRead = mBlockWriter.write(mAppBuffer.get(),
+                                               framesRead * mFilterStream->getBytesPerFrame());
+        if (bytesRead < 0) return bytesRead; // TODO review
+    }
+    return numFrames;
+}
+
+int32_t DataConversionFlowGraph::onProcessFixedBlock(uint8_t *buffer, int32_t numBytes) {
+    int32_t numFrames = numBytes / mFilterStream->getBytesPerFrame();
+    mCallbackResult = mFilterStream->getDataCallback()->onAudioReady(mFilterStream, buffer, numFrames);
+    // TODO handle STOP from callback, process data remaining in the block adapter
+    return numBytes;
+}

vendor/github.com/ebitengine/oto/v3/internal/oboe/oboe_common_DataConversionFlowGraph_android.h

@@ -0,0 +1,86 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef OBOE_OBOE_FLOW_GRAPH_H
+#define OBOE_OBOE_FLOW_GRAPH_H
+
+#include <memory>
+#include <stdint.h>
+#include <sys/types.h>
+
+#include "oboe_flowgraph_ChannelCountConverter_android.h"
+#include "oboe_flowgraph_MonoToMultiConverter_android.h"
+#include "oboe_flowgraph_MultiToMonoConverter_android.h"
+#include "oboe_flowgraph_SampleRateConverter_android.h"
+#include "oboe_oboe_Definitions_android.h"
+#include "oboe_common_AudioSourceCaller_android.h"
+#include "oboe_common_FixedBlockWriter_android.h"
+
+namespace oboe {
+
+class AudioStream;
+class AudioSourceCaller;
+
+/**
+ * Convert PCM channels, format and sample rate for optimal latency.
+ */
+class DataConversionFlowGraph : public FixedBlockProcessor {
+public:
+
+    DataConversionFlowGraph()
+    : mBlockWriter(*this) {}
+
+    void setSource(const void *buffer, int32_t numFrames);
+
+    /** Connect several modules together to convert from source to sink.
+     * This should only be called once for each instance.
+     *
+     * @param sourceFormat
+     * @param sourceChannelCount
+     * @param sinkFormat
+     * @param sinkChannelCount
+     * @return
+     */
+    oboe::Result configure(oboe::AudioStream *sourceStream, oboe::AudioStream *sinkStream);
+
+    int32_t read(void *buffer, int32_t numFrames, int64_t timeoutNanos);
+
+    int32_t write(void *buffer, int32_t numFrames);
+
+    int32_t onProcessFixedBlock(uint8_t *buffer, int32_t numBytes) override;
+
+    DataCallbackResult getDataCallbackResult() {
+        return mCallbackResult;
+    }
+
+private:
+    std::unique_ptr<flowgraph::FlowGraphSourceBuffered>    mSource;
+    std::unique_ptr<AudioSourceCaller>                 mSourceCaller;
+    std::unique_ptr<flowgraph::MonoToMultiConverter>   mMonoToMultiConverter;
+    std::unique_ptr<flowgraph::MultiToMonoConverter>   mMultiToMonoConverter;
+    std::unique_ptr<flowgraph::ChannelCountConverter>  mChannelCountConverter;
+    std::unique_ptr<resampler::MultiChannelResampler>  mResampler;
+    std::unique_ptr<flowgraph::SampleRateConverter>    mRateConverter;
+    std::unique_ptr<flowgraph::FlowGraphSink>              mSink;
+
+    FixedBlockWriter                                   mBlockWriter;
+    DataCallbackResult                                 mCallbackResult = DataCallbackResult::Continue;
+    AudioStream                                       *mFilterStream = nullptr;
+    std::unique_ptr<uint8_t[]>                         mAppBuffer;
+};
+
+}
+#endif //OBOE_OBOE_FLOW_GRAPH_H

vendor/github.com/ebitengine/oto/v3/internal/oboe/oboe_common_FilterAudioStream_android.cpp

@@ -0,0 +1,106 @@
+/*
+ * Copyright 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <memory>
+
+#include "oboe_common_OboeDebug_android.h"
+#include "oboe_common_FilterAudioStream_android.h"
+
+using namespace oboe;
+using namespace flowgraph;
+
+// Output callback uses FixedBlockReader::read()
+//                <= SourceFloatCaller::onProcess()
+//                <=== DataConversionFlowGraph::read()
+//                <== FilterAudioStream::onAudioReady()
+//
+// Output blocking uses no block adapter because AAudio can accept
+// writes of any size. It uses DataConversionFlowGraph::read() <== FilterAudioStream::write() <= app
+//
+// Input callback uses FixedBlockWriter::write()
+//                <= DataConversionFlowGraph::write()
+//                <= FilterAudioStream::onAudioReady()
+//
+// Input blocking uses FixedBlockReader::read() // TODO may not need block adapter
+//                <= SourceFloatCaller::onProcess()
+//                <=== SinkFloat::read()
+//                <= DataConversionFlowGraph::read()
+//                <== FilterAudioStream::read()
+//                <= app
+
+Result FilterAudioStream::configureFlowGraph() {
+    mFlowGraph = std::make_unique<DataConversionFlowGraph>();
+    bool isOutput = getDirection() == Direction::Output;
+
+    AudioStream *sourceStream =  isOutput ? this : mChildStream.get();
+    AudioStream *sinkStream =  isOutput ? mChildStream.get() : this;
+
+    mRateScaler = ((double) getSampleRate()) / mChildStream->getSampleRate();
+
+    return mFlowGraph->configure(sourceStream, sinkStream);
+}
+
+// Put the data to be written at the source end of the flowgraph.
+// Then read (pull) the data from the flowgraph and write it to the
+// child stream.
+ResultWithValue<int32_t> FilterAudioStream::write(const void *buffer,
+                               int32_t numFrames,
+                               int64_t timeoutNanoseconds) {
+    int32_t framesWritten = 0;
+    mFlowGraph->setSource(buffer, numFrames);
+    while (true) {
+        int32_t numRead = mFlowGraph->read(mBlockingBuffer.get(),
+                getFramesPerBurst(),
+                timeoutNanoseconds);
+        if (numRead < 0) {
+            return ResultWithValue<int32_t>::createBasedOnSign(numRead);
+        }
+        if (numRead == 0) {
+            break; // finished processing the source buffer
+        }
+        auto writeResult = mChildStream->write(mBlockingBuffer.get(),
+                                               numRead,
+                                               timeoutNanoseconds);
+        if (!writeResult) {
+            return writeResult;
+        }
+        framesWritten += writeResult.value();
+    }
+    return ResultWithValue<int32_t>::createBasedOnSign(framesWritten);
+}
+
+// Read (pull) the data we want from the sink end of the flowgraph.
+// The necessary data will be read from the child stream using a flowgraph callback.
+ResultWithValue<int32_t> FilterAudioStream::read(void *buffer,
+                                                  int32_t numFrames,
+                                                  int64_t timeoutNanoseconds) {
+    int32_t framesRead = mFlowGraph->read(buffer, numFrames, timeoutNanoseconds);
+    return ResultWithValue<int32_t>::createBasedOnSign(framesRead);
+}
+
+DataCallbackResult FilterAudioStream::onAudioReady(AudioStream *oboeStream,
+                                void *audioData,
+                                int32_t numFrames) {
+    int32_t framesProcessed;
+    if (oboeStream->getDirection() == Direction::Output) {
+        framesProcessed = mFlowGraph->read(audioData, numFrames, 0 /* timeout */);
+    } else {
+        framesProcessed = mFlowGraph->write(audioData, numFrames);
+    }
+    return (framesProcessed < numFrames)
+           ? DataCallbackResult::Stop
+           : mFlowGraph->getDataCallbackResult();
+}

vendor/github.com/ebitengine/oto/v3/internal/oboe/oboe_common_FilterAudioStream_android.h

@@ -0,0 +1,223 @@
+/*
+ * Copyright 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef OBOE_FILTER_AUDIO_STREAM_H
+#define OBOE_FILTER_AUDIO_STREAM_H
+
+#include <memory>
+#include "oboe_oboe_AudioStream_android.h"
+#include "oboe_common_DataConversionFlowGraph_android.h"
+
+namespace oboe {
+
+/**
+ * An AudioStream that wraps another AudioStream and provides audio data conversion.
+ * Operations may include channel conversion, data format conversion and/or sample rate conversion.
+ */
+class FilterAudioStream : public AudioStream, AudioStreamCallback {
+public:
+
+    /**
+     * Construct an `AudioStream` using the given `AudioStreamBuilder` and a child AudioStream.
+     *
+     * This should only be called internally by AudioStreamBuilder.
+     * Ownership of childStream will be passed to this object.
+     *
+     * @param builder containing all the stream's attributes
+     */
+    FilterAudioStream(const AudioStreamBuilder &builder, AudioStream *childStream)
+    : AudioStream(builder)
+    , mChildStream(childStream) {
+        // Intercept the callback if used.
+        if (builder.isErrorCallbackSpecified()) {
+            mErrorCallback = mChildStream->swapErrorCallback(this);
+        }
+        if (builder.isDataCallbackSpecified()) {
+            mDataCallback = mChildStream->swapDataCallback(this);
+        } else {
+            const int size = childStream->getFramesPerBurst() * childStream->getBytesPerFrame();
+            mBlockingBuffer = std::make_unique<uint8_t[]>(size);
+        }
+
+        // Copy parameters that may not match builder.
+        mBufferCapacityInFrames = mChildStream->getBufferCapacityInFrames();
+        mPerformanceMode = mChildStream->getPerformanceMode();
+        mInputPreset = mChildStream->getInputPreset();
+        mFramesPerBurst = mChildStream->getFramesPerBurst();
+        mDeviceId = mChildStream->getDeviceId();
+    }
+
+    virtual ~FilterAudioStream() = default;
+
+    AudioStream *getChildStream() const {
+        return mChildStream.get();
+    }
+
+    Result configureFlowGraph();
+
+    // Close child and parent.
+    Result close()  override {
+        const Result result1 = mChildStream->close();
+        const Result result2 = AudioStream::close();
+        return (result1 != Result::OK ? result1 : result2);
+    }
+
+    /**
+     * Start the stream asynchronously. Returns immediately (does not block). Equivalent to calling
+     * `start(0)`.
+     */
+    Result requestStart() override {
+        return mChildStream->requestStart();
+    }
+
+    /**
+     * Pause the stream asynchronously. Returns immediately (does not block). Equivalent to calling
+     * `pause(0)`.
+     */
+    Result requestPause() override {
+        return mChildStream->requestPause();
+    }
+
+    /**
+     * Flush the stream asynchronously. Returns immediately (does not block). Equivalent to calling
+     * `flush(0)`.
+     */
+    Result requestFlush() override {
+        return mChildStream->requestFlush();
+    }
+
+    /**
+     * Stop the stream asynchronously. Returns immediately (does not block). Equivalent to calling
+     * `stop(0)`.
+     */
+    Result requestStop() override {
+        return mChildStream->requestStop();
+    }
+
+    ResultWithValue<int32_t> read(void *buffer,
+            int32_t numFrames,
+            int64_t timeoutNanoseconds) override;
+
+    ResultWithValue<int32_t> write(const void *buffer,
+            int32_t numFrames,
+            int64_t timeoutNanoseconds) override;
+
+    StreamState getState() override {
+        return mChildStream->getState();
+    }
+
+    Result waitForStateChange(
+            StreamState inputState,
+            StreamState *nextState,
+            int64_t timeoutNanoseconds) override {
+        return mChildStream->waitForStateChange(inputState, nextState, timeoutNanoseconds);
+    }
+
+    bool isXRunCountSupported() const override {
+        return mChildStream->isXRunCountSupported();
+    }
+
+    AudioApi getAudioApi() const override {
+        return mChildStream->getAudioApi();
+    }
+
+    void updateFramesWritten() override {
+        // TODO for output, just count local writes?
+        mFramesWritten = static_cast<int64_t>(mChildStream->getFramesWritten() * mRateScaler);
+    }
+
+    void updateFramesRead() override {
+        // TODO for input, just count local reads?
+        mFramesRead = static_cast<int64_t>(mChildStream->getFramesRead() * mRateScaler);
+    }
+
+    void *getUnderlyingStream() const  override {
+        return mChildStream->getUnderlyingStream();
+    }
+
+    ResultWithValue<int32_t> setBufferSizeInFrames(int32_t requestedFrames) override {
+        return mChildStream->setBufferSizeInFrames(requestedFrames);
+    }
+
+    int32_t getBufferSizeInFrames() override {
+        mBufferSizeInFrames = mChildStream->getBufferSizeInFrames();
+        return mBufferSizeInFrames;
+    }
+
+    ResultWithValue<int32_t> getXRunCount() override {
+        return mChildStream->getXRunCount();
+    }
+
+    ResultWithValue<double> calculateLatencyMillis() override {
+        // This will automatically include the latency of the flowgraph?
+        return mChildStream->calculateLatencyMillis();
+    }
+
+    Result getTimestamp(clockid_t clockId,
+            int64_t *framePosition,
+            int64_t *timeNanoseconds) override {
+        int64_t childPosition = 0;
+        Result result = mChildStream->getTimestamp(clockId, &childPosition, timeNanoseconds);
+        // It is OK if framePosition is null.
+        if (framePosition) {
+            *framePosition = childPosition * mRateScaler;
+        }
+        return result;
+    }
+
+    DataCallbackResult onAudioReady(AudioStream *oboeStream,
+            void *audioData,
+            int32_t numFrames) override;
+
+    bool onError(AudioStream * /*audioStream*/, Result error) override {
+        if (mErrorCallback != nullptr) {
+            return mErrorCallback->onError(this, error);
+        }
+        return false;
+    }
+
+    void onErrorBeforeClose(AudioStream * /*oboeStream*/, Result error) override {
+        if (mErrorCallback != nullptr) {
+            mErrorCallback->onErrorBeforeClose(this, error);
+        }
+    }
+
+    void onErrorAfterClose(AudioStream * /*oboeStream*/, Result error) override {
+        // Close this parent stream because the callback will only close the child.
+        AudioStream::close();
+        if (mErrorCallback != nullptr) {
+            mErrorCallback->onErrorAfterClose(this, error);
+        }
+    }
+
+    /**
+     * @return last result passed from an error callback
+     */
+    oboe::Result getLastErrorCallbackResult() const override {
+        return mChildStream->getLastErrorCallbackResult();
+    }
+
+private:
+
+    std::unique_ptr<AudioStream>             mChildStream; // this stream wraps the child stream
+    std::unique_ptr<DataConversionFlowGraph> mFlowGraph; // for converting data
+    std::unique_ptr<uint8_t[]>               mBlockingBuffer; // temp buffer for write()
+    double                                   mRateScaler = 1.0; // ratio parent/child sample rates
+};
+
+} // oboe
+
+#endif //OBOE_FILTER_AUDIO_STREAM_H

vendor/github.com/ebitengine/oto/v3/internal/oboe/oboe_common_FixedBlockAdapter_android.cpp

@@ -0,0 +1,38 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <stdint.h>
+
+#include "oboe_common_FixedBlockAdapter_android.h"
+
+FixedBlockAdapter::~FixedBlockAdapter() {
+}
+
+int32_t FixedBlockAdapter::open(int32_t bytesPerFixedBlock)
+{
+    mSize = bytesPerFixedBlock;
+    mStorage = std::make_unique<uint8_t[]>(bytesPerFixedBlock);
+    mPosition = 0;
+    return 0;
+}
+
+int32_t FixedBlockAdapter::close()
+{
+    mStorage.reset(nullptr);
+    mSize = 0;
+    mPosition = 0;
+    return 0;
+}

vendor/github.com/ebitengine/oto/v3/internal/oboe/oboe_common_FixedBlockAdapter_android.h

@@ -0,0 +1,67 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef AAUDIO_FIXED_BLOCK_ADAPTER_H
+#define AAUDIO_FIXED_BLOCK_ADAPTER_H
+
+#include <memory>
+#include <stdint.h>
+#include <sys/types.h>
+
+/**
+ * Interface for a class that needs fixed-size blocks.
+ */
+class FixedBlockProcessor {
+public:
+    virtual ~FixedBlockProcessor() = default;
+    /**
+     *
+     * @param buffer Pointer to first byte of data.
+     * @param numBytes This will be a fixed size specified in FixedBlockAdapter::open().
+     * @return Number of bytes processed or a negative error code.
+     */
+    virtual int32_t onProcessFixedBlock(uint8_t *buffer, int32_t numBytes) = 0;
+};
+
+/**
+ * Base class for a variable-to-fixed-size block adapter.
+ */
+class FixedBlockAdapter
+{
+public:
+    FixedBlockAdapter(FixedBlockProcessor &fixedBlockProcessor)
+            : mFixedBlockProcessor(fixedBlockProcessor) {}
+
+    virtual ~FixedBlockAdapter();
+
+    /**
+     * Allocate internal resources needed for buffering data.
+     */
+    virtual int32_t open(int32_t bytesPerFixedBlock);
+
+    /**
+     * Free internal resources.
+     */
+    int32_t close();
+
+protected:
+    FixedBlockProcessor  &mFixedBlockProcessor;
+    std::unique_ptr<uint8_t[]> mStorage;       // Store data here while assembling buffers.
+    int32_t               mSize = 0;           // Size in bytes of the fixed size buffer.
+    int32_t               mPosition = 0;       // Offset of the last byte read or written.
+};
+
+#endif /* AAUDIO_FIXED_BLOCK_ADAPTER_H */

vendor/github.com/ebitengine/oto/v3/internal/oboe/oboe_common_FixedBlockReader_android.cpp

@@ -0,0 +1,73 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <stdint.h>
+#include <memory.h>
+
+#include "oboe_common_FixedBlockAdapter_android.h"
+
+#include "oboe_common_FixedBlockReader_android.h"
+
+
+FixedBlockReader::FixedBlockReader(FixedBlockProcessor &fixedBlockProcessor)
+        : FixedBlockAdapter(fixedBlockProcessor) {
+    mPosition = mSize;
+}
+
+int32_t FixedBlockReader::open(int32_t bytesPerFixedBlock) {
+    int32_t result = FixedBlockAdapter::open(bytesPerFixedBlock);
+    mPosition = 0;
+    mValid = 0;
+    return result;
+}
+
+int32_t FixedBlockReader::readFromStorage(uint8_t *buffer, int32_t numBytes) {
+    int32_t bytesToRead = numBytes;
+    int32_t dataAvailable = mValid - mPosition;
+    if (bytesToRead > dataAvailable) {
+        bytesToRead = dataAvailable;
+    }
+    memcpy(buffer, mStorage.get() + mPosition, bytesToRead);
+    mPosition += bytesToRead;
+    return bytesToRead;
+}
+
+int32_t FixedBlockReader::read(uint8_t *buffer, int32_t numBytes) {
+    int32_t bytesRead;
+    int32_t bytesLeft = numBytes;
+    while(bytesLeft > 0) {
+        if (mPosition < mValid) {
+            // Use up bytes currently in storage.
+            bytesRead = readFromStorage(buffer, bytesLeft);
+            buffer += bytesRead;
+            bytesLeft -= bytesRead;
+        } else if (bytesLeft >= mSize) {
+            // Nothing in storage. Read through if enough for a complete block.
+            bytesRead = mFixedBlockProcessor.onProcessFixedBlock(buffer, mSize);
+            if (bytesRead < 0) return bytesRead;
+            buffer += bytesRead;
+            bytesLeft -= bytesRead;
+        } else {
+            // Just need a partial block so we have to reload storage.
+            bytesRead = mFixedBlockProcessor.onProcessFixedBlock(mStorage.get(), mSize);
+            if (bytesRead < 0) return bytesRead;
+            mPosition = 0;
+            mValid = bytesRead;
+            if (bytesRead == 0) break;
+        }
+    }
+    return numBytes - bytesLeft;
+}

vendor/github.com/ebitengine/oto/v3/internal/oboe/oboe_common_FixedBlockReader_android.h

@@ -0,0 +1,60 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef AAUDIO_FIXED_BLOCK_READER_H
+#define AAUDIO_FIXED_BLOCK_READER_H
+
+#include <stdint.h>
+
+#include "oboe_common_FixedBlockAdapter_android.h"
+
+/**
+ * Read from a fixed-size block to a variable sized block.
+ *
+ * This can be used to convert a pull data flow from fixed sized buffers to variable sized buffers.
+ * An example would be an audio output callback that reads from the app.
+ */
+class FixedBlockReader : public FixedBlockAdapter
+{
+public:
+    FixedBlockReader(FixedBlockProcessor &fixedBlockProcessor);
+
+    virtual ~FixedBlockReader() = default;
+
+    int32_t open(int32_t bytesPerFixedBlock) override;
+
+    /**
+     * Read into a variable sized block.
+     *
+     * Note that if the fixed-sized blocks must be aligned, then the variable-sized blocks
+     * must have the same alignment.
+     * For example, if the fixed-size blocks must be a multiple of 8, then the variable-sized
+     * blocks must also be a multiple of 8.
+     *
+     * @param buffer
+     * @param numBytes
+     * @return Number of bytes read or a negative error code.
+     */
+    int32_t read(uint8_t *buffer, int32_t numBytes);
+
+private:
+    int32_t readFromStorage(uint8_t *buffer, int32_t numBytes);
+
+    int32_t               mValid = 0;            // Number of valid bytes in mStorage.
+};
+
+
+#endif /* AAUDIO_FIXED_BLOCK_READER_H */

vendor/github.com/ebitengine/oto/v3/internal/oboe/oboe_common_FixedBlockWriter_android.cpp

@@ -0,0 +1,73 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <stdint.h>
+#include <memory.h>
+
+#include "oboe_common_FixedBlockAdapter_android.h"
+#include "oboe_common_FixedBlockWriter_android.h"
+
+FixedBlockWriter::FixedBlockWriter(FixedBlockProcessor &fixedBlockProcessor)
+        : FixedBlockAdapter(fixedBlockProcessor) {}
+
+
+int32_t FixedBlockWriter::writeToStorage(uint8_t *buffer, int32_t numBytes) {
+    int32_t bytesToStore = numBytes;
+    int32_t roomAvailable = mSize - mPosition;
+    if (bytesToStore > roomAvailable) {
+        bytesToStore = roomAvailable;
+    }
+    memcpy(mStorage.get() + mPosition, buffer, bytesToStore);
+    mPosition += bytesToStore;
+    return bytesToStore;
+}
+
+int32_t FixedBlockWriter::write(uint8_t *buffer, int32_t numBytes) {
+    int32_t bytesLeft = numBytes;
+
+    // If we already have data in storage then add to it.
+    if (mPosition > 0) {
+        int32_t bytesWritten = writeToStorage(buffer, bytesLeft);
+        buffer += bytesWritten;
+        bytesLeft -= bytesWritten;
+        // If storage full then flush it out
+        if (mPosition == mSize) {
+            bytesWritten = mFixedBlockProcessor.onProcessFixedBlock(mStorage.get(), mSize);
+            if (bytesWritten < 0) return bytesWritten;
+            mPosition = 0;
+            if (bytesWritten < mSize) {
+                // Only some of the data was written! This should not happen.
+                return -1;
+            }
+        }
+    }
+
+    // Write through if enough for a complete block.
+    while(bytesLeft > mSize) {
+        int32_t bytesWritten = mFixedBlockProcessor.onProcessFixedBlock(buffer, mSize);
+        if (bytesWritten < 0) return bytesWritten;
+        buffer += bytesWritten;
+        bytesLeft -= bytesWritten;
+    }
+
+    // Save any remaining partial blocks for next time.
+    if (bytesLeft > 0) {
+        int32_t bytesWritten = writeToStorage(buffer, bytesLeft);
+        bytesLeft -= bytesWritten;
+    }
+
+    return numBytes - bytesLeft;
+}

vendor/github.com/ebitengine/oto/v3/internal/oboe/oboe_common_FixedBlockWriter_android.h

@@ -0,0 +1,54 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef AAUDIO_FIXED_BLOCK_WRITER_H
+#define AAUDIO_FIXED_BLOCK_WRITER_H
+
+#include <stdint.h>
+
+#include "oboe_common_FixedBlockAdapter_android.h"
+
+/**
+ * This can be used to convert a push data flow from variable sized buffers to fixed sized buffers.
+ * An example would be an audio input callback.
+ */
+class FixedBlockWriter : public FixedBlockAdapter
+{
+public:
+    FixedBlockWriter(FixedBlockProcessor &fixedBlockProcessor);
+
+    virtual ~FixedBlockWriter() = default;
+
+    /**
+     * Write from a variable sized block.
+     *
+     * Note that if the fixed-sized blocks must be aligned, then the variable-sized blocks
+     * must have the same alignment.
+     * For example, if the fixed-size blocks must be a multiple of 8, then the variable-sized
+     * blocks must also be a multiple of 8.
+     *
+     * @param buffer
+     * @param numBytes
+     * @return Number of bytes written or a negative error code.
+     */
+    int32_t write(uint8_t *buffer, int32_t numBytes);
+
+private:
+
+    int32_t writeToStorage(uint8_t *buffer, int32_t numBytes);
+};
+
+#endif /* AAUDIO_FIXED_BLOCK_WRITER_H */

vendor/github.com/ebitengine/oto/v3/internal/oboe/oboe_common_LatencyTuner_android.cpp

@@ -0,0 +1,108 @@
+/*
+ * Copyright 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include "oboe_oboe_LatencyTuner_android.h"
+
+using namespace oboe;
+
+LatencyTuner::LatencyTuner(AudioStream &stream)
+        : LatencyTuner(stream, stream.getBufferCapacityInFrames()) {
+}
+
+LatencyTuner::LatencyTuner(oboe::AudioStream &stream, int32_t maximumBufferSize)
+        : mStream(stream)
+        , mMaxBufferSize(maximumBufferSize) {
+    int32_t burstSize = stream.getFramesPerBurst();
+    setMinimumBufferSize(kDefaultNumBursts * burstSize);
+    setBufferSizeIncrement(burstSize);
+    reset();
+}
+
+Result LatencyTuner::tune() {
+    if (mState == State::Unsupported) {
+        return Result::ErrorUnimplemented;
+    }
+
+    Result result = Result::OK;
+
+    // Process reset requests.
+    int32_t numRequests = mLatencyTriggerRequests.load();
+    if (numRequests != mLatencyTriggerResponses.load()) {
+        mLatencyTriggerResponses.store(numRequests);
+        reset();
+    }
+
+    // Set state to Active if the idle countdown has reached zero.
+    if (mState == State::Idle && --mIdleCountDown <= 0) {
+        mState = State::Active;
+    }
+
+    // When state is Active attempt to change the buffer size if the number of xRuns has increased.
+    if (mState == State::Active) {
+
+        auto xRunCountResult = mStream.getXRunCount();
+        if (xRunCountResult == Result::OK) {
+            if ((xRunCountResult.value() - mPreviousXRuns) > 0) {
+                mPreviousXRuns = xRunCountResult.value();
+                int32_t oldBufferSize = mStream.getBufferSizeInFrames();
+                int32_t requestedBufferSize = oldBufferSize + getBufferSizeIncrement();
+
+                // Do not request more than the maximum buffer size (which was either user-specified
+                // or was from stream->getBufferCapacityInFrames())
+                if (requestedBufferSize > mMaxBufferSize) requestedBufferSize = mMaxBufferSize;
+
+                // Note that this will not allocate more memory. It simply determines
+                // how much of the existing buffer capacity will be used. The size will be
+                // clipped to the bufferCapacity by AAudio.
+                auto setBufferResult = mStream.setBufferSizeInFrames(requestedBufferSize);
+                if (setBufferResult != Result::OK) {
+                    result = setBufferResult;
+                    mState = State::Unsupported;
+                } else if (setBufferResult.value() == oldBufferSize) {
+                    mState = State::AtMax;
+                }
+            }
+        } else {
+            mState = State::Unsupported;
+        }
+    }
+
+    if (mState == State::Unsupported) {
+        result = Result::ErrorUnimplemented;
+    }
+
+    if (mState == State::AtMax) {
+        result = Result::OK;
+    }
+    return result;
+}
+
+void LatencyTuner::requestReset() {
+    if (mState != State::Unsupported) {
+        mLatencyTriggerRequests++;
+    }
+}
+
+void LatencyTuner::reset() {
+    mState = State::Idle;
+    mIdleCountDown = kIdleCount;
+    // Set to minimal latency
+    mStream.setBufferSizeInFrames(getMinimumBufferSize());
+}
+
+bool LatencyTuner::isAtMaximumBufferSize() {
+    return mState == State::AtMax;
+}

vendor/github.com/ebitengine/oto/v3/internal/oboe/oboe_common_MonotonicCounter_android.h

@@ -0,0 +1,112 @@
+/*
+ * Copyright 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef COMMON_MONOTONIC_COUNTER_H
+#define COMMON_MONOTONIC_COUNTER_H
+
+#include <cstdint>
+
+/**
+ * Maintain a 64-bit monotonic counter.
+ * Can be used to track a 32-bit counter that wraps or gets reset.
+ *
+ * Note that this is not atomic and has no interior locks.
+ * A caller will need to provide their own exterior locking
+ * if they need to use it from multiple threads.
+ */
+class MonotonicCounter {
+
+public:
+    MonotonicCounter() {}
+    virtual ~MonotonicCounter() {}
+
+    /**
+     * @return current value of the counter
+     */
+    int64_t get() const {
+        return mCounter64;
+    }
+
+    /**
+     * set the current value of the counter
+     */
+    void set(int64_t counter) {
+        mCounter64 = counter;
+    }
+
+    /**
+     * Advance the counter if delta is positive.
+     * @return current value of the counter
+     */
+    int64_t increment(int64_t delta) {
+        if (delta > 0) {
+            mCounter64 += delta;
+        }
+        return mCounter64;
+    }
+
+    /**
+     * Advance the 64-bit counter if (current32 - previousCurrent32) > 0.
+     * This can be used to convert a 32-bit counter that may be wrapping into
+     * a monotonic 64-bit counter.
+     *
+     * This counter32 should NOT be allowed to advance by more than 0x7FFFFFFF between calls.
+     * Think of the wrapping counter like a sine wave. If the frequency of the signal
+     * is more than half the sampling rate (Nyquist rate) then you cannot measure it properly.
+     * If the counter wraps around every 24 hours then we should measure it with a period
+     * of less than 12 hours.
+     *
+     * @return current value of the 64-bit counter
+     */
+    int64_t update32(int32_t counter32) {
+        int32_t delta = counter32 - mCounter32;
+        // protect against the mCounter64 going backwards
+        if (delta > 0) {
+            mCounter64 += delta;
+            mCounter32 = counter32;
+        }
+        return mCounter64;
+    }
+
+    /**
+     * Reset the stored value of the 32-bit counter.
+     * This is used if your counter32 has been reset to zero.
+     */
+    void reset32() {
+        mCounter32 = 0;
+    }
+
+    /**
+     * Round 64-bit counter up to a multiple of the period.
+     *
+     * The period must be positive.
+     *
+     * @param period might be, for example, a buffer capacity
+     */
+    void roundUp64(int32_t period) {
+        if (period > 0) {
+            int64_t numPeriods = (mCounter64 + period - 1) / period;
+            mCounter64 = numPeriods * period;
+        }
+    }
+
+private:
+    int64_t mCounter64 = 0;
+    int32_t mCounter32 = 0;
+};
+
+
+#endif //COMMON_MONOTONIC_COUNTER_H

vendor/github.com/ebitengine/oto/v3/internal/oboe/oboe_common_OboeDebug_android.h

@@ -0,0 +1,41 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ */
+
+#ifndef OBOE_DEBUG_H
+#define OBOE_DEBUG_H
+
+#include <android/log.h>
+
+#ifndef MODULE_NAME
+#define MODULE_NAME  "OboeAudio"
+#endif
+
+// Always log INFO and errors.
+#define LOGI(...) __android_log_print(ANDROID_LOG_INFO, MODULE_NAME, __VA_ARGS__)
+#define LOGW(...) __android_log_print(ANDROID_LOG_WARN, MODULE_NAME, __VA_ARGS__)
+#define LOGE(...) __android_log_print(ANDROID_LOG_ERROR, MODULE_NAME, __VA_ARGS__)
+#define LOGF(...) __android_log_print(ANDROID_LOG_FATAL, MODULE_NAME, __VA_ARGS__)
+
+#if OBOE_ENABLE_LOGGING
+#define LOGV(...) __android_log_print(ANDROID_LOG_VERBOSE, MODULE_NAME, __VA_ARGS__)
+#define LOGD(...) __android_log_print(ANDROID_LOG_DEBUG, MODULE_NAME, __VA_ARGS__)
+#else
+#define LOGV(...)
+#define LOGD(...)
+#endif
+
+#endif //OBOE_DEBUG_H

vendor/github.com/ebitengine/oto/v3/internal/oboe/oboe_common_OboeExtensions_android.cpp

@@ -0,0 +1,36 @@
+/*
+ * Copyright (C) 2022 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include "oboe_oboe_OboeExtensions_android.h"
+#include "oboe_aaudio_AAudioExtensions_android.h"
+
+using namespace oboe;
+
+bool OboeExtensions::isMMapSupported(){
+    return AAudioExtensions::getInstance().isMMapSupported();
+}
+
+bool OboeExtensions::isMMapEnabled(){
+    return AAudioExtensions::getInstance().isMMapEnabled();
+}
+
+int32_t OboeExtensions::setMMapEnabled(bool enabled){
+    return AAudioExtensions::getInstance().setMMapEnabled(enabled);
+}
+
+bool OboeExtensions::isMMapUsed(oboe::AudioStream *oboeStream){
+    return AAudioExtensions::getInstance().isMMapUsed(oboeStream);
+}

vendor/github.com/ebitengine/oto/v3/internal/oboe/oboe_common_QuirksManager_android.cpp

@@ -0,0 +1,304 @@
+/*
+ * Copyright 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include "oboe_oboe_AudioStreamBuilder_android.h"
+#include "oboe_oboe_Oboe_android.h"
+
+#include "oboe_common_OboeDebug_android.h"
+#include "oboe_common_QuirksManager_android.h"
+
+using namespace oboe;
+
+int32_t QuirksManager::DeviceQuirks::clipBufferSize(AudioStream &stream,
+            int32_t requestedSize) {
+    if (!OboeGlobals::areWorkaroundsEnabled()) {
+        return requestedSize;
+    }
+    int bottomMargin = kDefaultBottomMarginInBursts;
+    int topMargin = kDefaultTopMarginInBursts;
+    if (isMMapUsed(stream)) {
+        if (stream.getSharingMode() == SharingMode::Exclusive) {
+            bottomMargin = getExclusiveBottomMarginInBursts();
+            topMargin = getExclusiveTopMarginInBursts();
+        }
+    } else {
+        bottomMargin = kLegacyBottomMarginInBursts;
+    }
+
+    int32_t burst = stream.getFramesPerBurst();
+    int32_t minSize = bottomMargin * burst;
+    int32_t adjustedSize = requestedSize;
+    if (adjustedSize < minSize ) {
+        adjustedSize = minSize;
+    } else {
+        int32_t maxSize = stream.getBufferCapacityInFrames() - (topMargin * burst);
+        if (adjustedSize > maxSize ) {
+            adjustedSize = maxSize;
+        }
+    }
+    return adjustedSize;
+}
+
+bool QuirksManager::DeviceQuirks::isAAudioMMapPossible(const AudioStreamBuilder &builder) const {
+    bool isSampleRateCompatible =
+            builder.getSampleRate() == oboe::Unspecified
+            || builder.getSampleRate() == kCommonNativeRate
+            || builder.getSampleRateConversionQuality() != SampleRateConversionQuality::None;
+    return builder.getPerformanceMode() == PerformanceMode::LowLatency
+            && isSampleRateCompatible
+            && builder.getChannelCount() <= kChannelCountStereo;
+}
+
+bool QuirksManager::DeviceQuirks::shouldConvertFloatToI16ForOutputStreams() {
+    std::string productManufacturer = getPropertyString("ro.product.manufacturer");
+    if (getSdkVersion() < __ANDROID_API_L__) {
+        return true;
+    } else if ((productManufacturer == "vivo") && (getSdkVersion() < __ANDROID_API_M__)) {
+        return true;
+    }
+    return false;
+}
+
+/**
+ * This is for Samsung Exynos quirks. Samsung Mobile uses Qualcomm chips so
+ * the QualcommDeviceQuirks would apply.
+ */
+class SamsungExynosDeviceQuirks : public  QuirksManager::DeviceQuirks {
+public:
+    SamsungExynosDeviceQuirks() {
+        std::string chipname = getPropertyString("ro.hardware.chipname");
+        isExynos9810 = (chipname == "exynos9810");
+        isExynos990 = (chipname == "exynos990");
+        isExynos850 = (chipname == "exynos850");
+
+        mBuildChangelist = getPropertyInteger("ro.build.changelist", 0);
+    }
+
+    virtual ~SamsungExynosDeviceQuirks() = default;
+
+    int32_t getExclusiveBottomMarginInBursts() const override {
+        return kBottomMargin;
+    }
+
+    int32_t getExclusiveTopMarginInBursts() const override {
+        return kTopMargin;
+    }
+
+    // See Oboe issues #824 and #1247 for more information.
+    bool isMonoMMapActuallyStereo() const override {
+        return isExynos9810 || isExynos850; // TODO We can make this version specific if it gets fixed.
+    }
+
+    bool isAAudioMMapPossible(const AudioStreamBuilder &builder) const override {
+        return DeviceQuirks::isAAudioMMapPossible(builder)
+                // Samsung says they use Legacy for Camcorder
+                && builder.getInputPreset() != oboe::InputPreset::Camcorder;
+    }
+
+    bool isMMapSafe(const AudioStreamBuilder &builder) override {
+        const bool isInput = builder.getDirection() == Direction::Input;
+        // This detects b/159066712 , S20 LSI has corrupt low latency audio recording
+        // and turns off MMAP.
+        // See also https://github.com/google/oboe/issues/892
+        bool isRecordingCorrupted = isInput
+            && isExynos990
+            && mBuildChangelist < 19350896;
+
+        // Certain S9+ builds record silence when using MMAP and not using the VoiceCommunication
+        // preset.
+        // See https://github.com/google/oboe/issues/1110
+        bool wouldRecordSilence = isInput
+            && isExynos9810
+            && mBuildChangelist <= 18847185
+            && (builder.getInputPreset() != InputPreset::VoiceCommunication);
+
+        if (wouldRecordSilence){
+            LOGI("QuirksManager::%s() Requested stream configuration would result in silence on "
+                 "this device. Switching off MMAP.", __func__);
+        }
+
+        return !isRecordingCorrupted && !wouldRecordSilence;
+    }
+
+private:
+    // Stay farther away from DSP position on Exynos devices.
+    static constexpr int32_t kBottomMargin = 2;
+    static constexpr int32_t kTopMargin = 1;
+    bool isExynos9810 = false;
+    bool isExynos990 = false;
+    bool isExynos850 = false;
+    int mBuildChangelist = 0;
+};
+
+class QualcommDeviceQuirks : public  QuirksManager::DeviceQuirks {
+public:
+    QualcommDeviceQuirks() {
+        std::string modelName = getPropertyString("ro.soc.model");
+        isSM8150 = (modelName == "SDM8150");
+    }
+
+    virtual ~QualcommDeviceQuirks() = default;
+
+    int32_t getExclusiveBottomMarginInBursts() const override {
+        return kBottomMargin;
+    }
+
+    bool isMMapSafe(const AudioStreamBuilder &builder) override {
+        // See https://github.com/google/oboe/issues/1121#issuecomment-897957749
+        bool isMMapBroken = false;
+        if (isSM8150 && (getSdkVersion() <= __ANDROID_API_P__)) {
+            LOGI("QuirksManager::%s() MMAP not actually supported on this chip."
+                 " Switching off MMAP.", __func__);
+            isMMapBroken = true;
+        }
+
+        return !isMMapBroken;
+    }
+
+private:
+    bool isSM8150 = false;
+    static constexpr int32_t kBottomMargin = 1;
+};
+
+QuirksManager::QuirksManager() {
+    std::string productManufacturer = getPropertyString("ro.product.manufacturer");
+    if (productManufacturer == "samsung") {
+        std::string arch = getPropertyString("ro.arch");
+        bool isExynos = (arch.rfind("exynos", 0) == 0); // starts with?
+        if (isExynos) {
+            mDeviceQuirks = std::make_unique<SamsungExynosDeviceQuirks>();
+        }
+    }
+    if (!mDeviceQuirks) {
+        std::string socManufacturer = getPropertyString("ro.soc.manufacturer");
+        if (socManufacturer == "Qualcomm") {
+            // This may include Samsung Mobile devices.
+            mDeviceQuirks = std::make_unique<QualcommDeviceQuirks>();
+        } else {
+            mDeviceQuirks = std::make_unique<DeviceQuirks>();
+        }
+    }
+}
+
+bool QuirksManager::isConversionNeeded(
+        const AudioStreamBuilder &builder,
+        AudioStreamBuilder &childBuilder) {
+    bool conversionNeeded = false;
+    const bool isLowLatency = builder.getPerformanceMode() == PerformanceMode::LowLatency;
+    const bool isInput = builder.getDirection() == Direction::Input;
+    const bool isFloat = builder.getFormat() == AudioFormat::Float;
+
+    // There are multiple bugs involving using callback with a specified callback size.
+    // Issue #778: O to Q had a problem with Legacy INPUT streams for FLOAT streams
+    // and a specified callback size. It would assert because of a bad buffer size.
+    //
+    // Issue #973: O to R had a problem with Legacy output streams using callback and a specified callback size.
+    // An AudioTrack stream could still be running when the AAudio FixedBlockReader was closed.
+    // Internally b/161914201#comment25
+    //
+    // Issue #983: O to R would glitch if the framesPerCallback was too small.
+    //
+    // Most of these problems were related to Legacy stream. MMAP was OK. But we don't
+    // know if we will get an MMAP stream. So, to be safe, just do the conversion in Oboe.
+    if (OboeGlobals::areWorkaroundsEnabled()
+            && builder.willUseAAudio()
+            && builder.isDataCallbackSpecified()
+            && builder.getFramesPerDataCallback() != 0
+            && getSdkVersion() <= __ANDROID_API_R__) {
+        LOGI("QuirksManager::%s() avoid setFramesPerCallback(n>0)", __func__);
+        childBuilder.setFramesPerCallback(oboe::Unspecified);
+        conversionNeeded = true;
+    }
+
+    // If a SAMPLE RATE is specified for low latency, let the native code choose an optimal rate.
+    // This isn't really a workaround. It is an Oboe feature that is convenient to place here.
+    // TODO There may be a problem if the devices supports low latency
+    //      at a higher rate than the default.
+    if (builder.getSampleRate() != oboe::Unspecified
+            && builder.getSampleRateConversionQuality() != SampleRateConversionQuality::None
+            && isLowLatency
+            ) {
+        childBuilder.setSampleRate(oboe::Unspecified); // native API decides the best sample rate
+        conversionNeeded = true;
+    }
+
+    // Data Format
+    // OpenSL ES and AAudio before P do not support FAST path for FLOAT capture.
+    if (OboeGlobals::areWorkaroundsEnabled()
+            && isFloat
+            && isInput
+            && builder.isFormatConversionAllowed()
+            && isLowLatency
+            && (!builder.willUseAAudio() || (getSdkVersion() < __ANDROID_API_P__))
+            ) {
+        childBuilder.setFormat(AudioFormat::I16); // needed for FAST track
+        conversionNeeded = true;
+        LOGI("QuirksManager::%s() forcing internal format to I16 for low latency", __func__);
+    }
+
+    // Add quirk for float output when needed.
+    if (OboeGlobals::areWorkaroundsEnabled()
+            && isFloat
+            && !isInput
+            && builder.isFormatConversionAllowed()
+            && mDeviceQuirks->shouldConvertFloatToI16ForOutputStreams()
+            ) {
+        childBuilder.setFormat(AudioFormat::I16);
+        conversionNeeded = true;
+        LOGI("QuirksManager::%s() float was requested but not supported on pre-L devices "
+             "and some devices like Vivo devices may have issues on L devices, "
+             "creating an underlying I16 stream and using format conversion to provide a float "
+             "stream", __func__);
+    }
+
+    // Channel Count conversions
+    if (OboeGlobals::areWorkaroundsEnabled()
+            && builder.isChannelConversionAllowed()
+            && builder.getChannelCount() == kChannelCountStereo
+            && isInput
+            && isLowLatency
+            && (!builder.willUseAAudio() && (getSdkVersion() == __ANDROID_API_O__))
+            ) {
+        // Workaround for heap size regression in O.
+        // b/66967812 AudioRecord does not allow FAST track for stereo capture in O
+        childBuilder.setChannelCount(kChannelCountMono);
+        conversionNeeded = true;
+        LOGI("QuirksManager::%s() using mono internally for low latency on O", __func__);
+    } else if (OboeGlobals::areWorkaroundsEnabled()
+               && builder.getChannelCount() == kChannelCountMono
+               && isInput
+               && mDeviceQuirks->isMonoMMapActuallyStereo()
+               && builder.willUseAAudio()
+               // Note: we might use this workaround on a device that supports
+               // MMAP but will use Legacy for this stream.  But this will only happen
+               // on devices that have the broken mono.
+               && mDeviceQuirks->isAAudioMMapPossible(builder)
+               ) {
+        // Workaround for mono actually running in stereo mode.
+        childBuilder.setChannelCount(kChannelCountStereo); // Use stereo and extract first channel.
+        conversionNeeded = true;
+        LOGI("QuirksManager::%s() using stereo internally to avoid broken mono", __func__);
+    }
+    // Note that MMAP does not support mono in 8.1. But that would only matter on Pixel 1
+    // phones and they have almost all been updated to 9.0.
+
+    return conversionNeeded;
+}
+
+bool QuirksManager::isMMapSafe(AudioStreamBuilder &builder) {
+    if (!OboeGlobals::areWorkaroundsEnabled()) return true;
+    return mDeviceQuirks->isMMapSafe(builder);
+}

vendor/github.com/ebitengine/oto/v3/internal/oboe/oboe_common_QuirksManager_android.h

@@ -0,0 +1,134 @@
+/*
+ * Copyright 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef OBOE_QUIRKS_MANAGER_H
+#define OBOE_QUIRKS_MANAGER_H
+
+#include <memory>
+#include "oboe_oboe_AudioStreamBuilder_android.h"
+#include "oboe_aaudio_AudioStreamAAudio_android.h"
+
+#ifndef __ANDROID_API_R__
+#define __ANDROID_API_R__ 30
+#endif
+
+namespace oboe {
+
+/**
+ * INTERNAL USE ONLY.
+ *
+ * Based on manufacturer, model and Android version number
+ * decide whether data conversion needs to occur.
+ *
+ * This also manages device and version specific workarounds.
+ */
+
+class QuirksManager {
+public:
+
+    static QuirksManager &getInstance() {
+        static QuirksManager instance; // singleton
+        return instance;
+    }
+
+    QuirksManager();
+    virtual ~QuirksManager() = default;
+
+    /**
+     * Do we need to do channel, format or rate conversion to provide a low latency
+     * stream for this builder? If so then provide a builder for the native child stream
+     * that will be used to get low latency.
+     *
+     * @param builder builder provided by application
+     * @param childBuilder modified builder appropriate for the underlying device
+     * @return true if conversion is needed
+     */
+    bool isConversionNeeded(const AudioStreamBuilder &builder, AudioStreamBuilder &childBuilder);
+
+    static bool isMMapUsed(AudioStream &stream) {
+        bool answer = false;
+        if (stream.getAudioApi() == AudioApi::AAudio) {
+            AudioStreamAAudio *streamAAudio =
+                    reinterpret_cast<AudioStreamAAudio *>(&stream);
+            answer = streamAAudio->isMMapUsed();
+        }
+        return answer;
+    }
+
+    virtual int32_t clipBufferSize(AudioStream &stream, int32_t bufferSize) {
+        return mDeviceQuirks->clipBufferSize(stream, bufferSize);
+    }
+
+    class DeviceQuirks {
+    public:
+        virtual ~DeviceQuirks() = default;
+
+        /**
+         * Restrict buffer size. This is mainly to avoid glitches caused by MMAP
+         * timestamp inaccuracies.
+         * @param stream
+         * @param requestedSize
+         * @return
+         */
+        int32_t clipBufferSize(AudioStream &stream, int32_t requestedSize);
+
+        // Exclusive MMAP streams can have glitches because they are using a timing
+        // model of the DSP to control IO instead of direct synchronization.
+        virtual int32_t getExclusiveBottomMarginInBursts() const {
+            return kDefaultBottomMarginInBursts;
+        }
+
+        virtual int32_t getExclusiveTopMarginInBursts() const {
+            return kDefaultTopMarginInBursts;
+        }
+
+        // On some devices, you can open a mono stream but it is actually running in stereo!
+        virtual bool isMonoMMapActuallyStereo() const {
+            return false;
+        }
+
+        virtual bool isAAudioMMapPossible(const AudioStreamBuilder &builder) const;
+
+        virtual bool isMMapSafe(const AudioStreamBuilder & /* builder */ ) {
+            return true;
+        }
+
+        // On some devices, Float does not work so it should be converted to I16.
+        static bool shouldConvertFloatToI16ForOutputStreams();
+
+        static constexpr int32_t kDefaultBottomMarginInBursts = 0;
+        static constexpr int32_t kDefaultTopMarginInBursts = 0;
+
+        // For Legacy streams, do not let the buffer go below one burst.
+        // b/129545119 | AAudio Legacy allows setBufferSizeInFrames too low
+        // Fixed in Q
+        static constexpr int32_t kLegacyBottomMarginInBursts = 1;
+        static constexpr int32_t kCommonNativeRate = 48000; // very typical native sample rate
+    };
+
+    bool isMMapSafe(AudioStreamBuilder &builder);
+
+private:
+
+    static constexpr int32_t kChannelCountMono = 1;
+    static constexpr int32_t kChannelCountStereo = 2;
+
+    std::unique_ptr<DeviceQuirks> mDeviceQuirks{};
+
+};
+
+}
+#endif //OBOE_QUIRKS_MANAGER_H