mirror of
https://github.com/hajimehoshi/ebiten.git
synced 2024-11-10 04:57:26 +01:00
539 lines
13 KiB
Go
539 lines
13 KiB
Go
// Copyright 2015 Hajime Hoshi
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//
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// Licensed under the Apache License, Version 2.0 (the "License");
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// you may not use this file except in compliance with the License.
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// You may obtain a copy of the License at
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//
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// http://www.apache.org/licenses/LICENSE-2.0
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//
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// Unless required by applicable law or agreed to in writing, software
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// distributed under the License is distributed on an "AS IS" BASIS,
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// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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// See the License for the specific language governing permissions and
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// limitations under the License.
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// Package audio provides audio players.
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//
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// The stream format must be 16-bit little endian and 2 channels. The format is as follows:
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// [data] = [sample 1] [sample 2] [sample 3] ...
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// [sample *] = [channel 1] ...
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// [channel *] = [byte 1] [byte 2] ...
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//
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// An audio context (audio.Context object) has a sample rate you can specify and all streams you want to play must have the same
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// sample rate. However, decoders in e.g. audio/mp3 package adjust sample rate automatically,
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// and you don't have to care about it as long as you use those decoders.
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//
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// An audio context can generate 'players' (audio.Player objects),
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// and you can play sound by calling Play function of players.
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// When multiple players play, mixing is automatically done.
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// Note that too many players may cause distortion.
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//
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// Ebiten's game progress always synchronizes with audio progress.
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//
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// For the simplest example to play sound, see wav package in the examples.
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package audio
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import (
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"bytes"
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"errors"
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"io"
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"runtime"
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"sync"
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"time"
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"github.com/hajimehoshi/oto"
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"github.com/hajimehoshi/ebiten/internal/audiobinding"
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"github.com/hajimehoshi/ebiten/internal/clock"
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"github.com/hajimehoshi/ebiten/internal/web"
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)
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type players struct {
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players map[*Player]struct{}
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sync.RWMutex
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}
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const (
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channelNum = 2
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bytesPerSample = 2
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// TODO: This assumes that channelNum is a power of 2.
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mask = ^(channelNum*bytesPerSample - 1)
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)
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func (p *players) Read(b []uint8) (int, error) {
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p.Lock()
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defer p.Unlock()
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players := []*Player{}
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for player := range p.players {
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players = append(players, player)
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}
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if len(players) == 0 {
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l := len(b)
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l &= mask
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copy(b, make([]uint8, l))
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return l, nil
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}
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closed := []*Player{}
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for _, player := range players {
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player.resetBufferIfSeeking()
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}
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l := len(b)
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for _, player := range players {
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n, err := player.readToBuffer(l)
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if err == io.EOF {
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closed = append(closed, player)
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} else if err != nil {
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return 0, err
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}
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if n < l {
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l = n
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}
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}
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l &= mask
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b16s := [][]int16{}
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for _, player := range players {
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b16s = append(b16s, player.bufferToInt16(l))
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}
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for i := 0; i < l/2; i++ {
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x := 0
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for _, b16 := range b16s {
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x += int(b16[i])
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}
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if x > (1<<15)-1 {
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x = (1 << 15) - 1
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}
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if x < -(1 << 15) {
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x = -(1 << 15)
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}
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b[2*i] = byte(x)
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b[2*i+1] = byte(x >> 8)
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}
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for _, player := range players {
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player.proceed(l)
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}
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for _, pl := range closed {
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delete(p.players, pl)
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}
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return l, nil
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}
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func (p *players) addPlayer(player *Player) {
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p.Lock()
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p.players[player] = struct{}{}
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p.Unlock()
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}
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func (p *players) removePlayer(player *Player) {
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p.Lock()
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delete(p.players, player)
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p.Unlock()
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}
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func (p *players) hasPlayer(player *Player) bool {
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p.RLock()
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_, ok := p.players[player]
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p.RUnlock()
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return ok
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}
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func (p *players) hasSource(src ReadSeekCloser) bool {
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p.RLock()
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defer p.RUnlock()
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for player := range p.players {
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if player.src == src {
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return true
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}
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}
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return false
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}
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// A Context represents a current state of audio.
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//
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// At most one Context object can exist in one process.
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// This means only one constant sample rate is valid in your one application.
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//
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// For a typical usage example, see examples/wav/main.go.
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type Context struct {
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players *players
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initCh chan struct{}
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initedCh chan struct{}
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pingCount int
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sampleRate int
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frames int64
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framesReadOnly int64
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writtenBytes int64
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m sync.Mutex
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}
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var (
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theContext *Context
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theContextLock sync.Mutex
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)
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// NewContext creates a new audio context with the given sample rate.
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//
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// The sample rate is also used for decoding MP3 with audio/mp3 package
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// or other formats as the target sample rate.
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//
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// sampleRate should be 44100 or 48000.
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// Other values might not work.
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// For example, 22050 causes error on Safari when decoding MP3.
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//
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// Error returned by NewContext is always nil as of 1.5.0-alpha.
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//
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// NewContext panics when an audio context is already created.
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func NewContext(sampleRate int) (*Context, error) {
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theContextLock.Lock()
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defer theContextLock.Unlock()
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if theContext != nil {
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panic("audio: context is already created")
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}
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c := &Context{
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sampleRate: sampleRate,
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}
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theContext = c
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c.players = &players{
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players: map[*Player]struct{}{},
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}
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go c.loop()
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return c, nil
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}
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// CurrentContext returns the current context or nil if there is no context.
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func CurrentContext() *Context {
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theContextLock.Lock()
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c := theContext
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theContextLock.Unlock()
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return c
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}
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func (c *Context) ping() {
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if c.initCh != nil {
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close(c.initCh)
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c.initCh = nil
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}
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<-c.initedCh
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c.m.Lock()
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c.pingCount = 5
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c.m.Unlock()
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}
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func (c *Context) loop() {
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c.initCh = make(chan struct{})
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c.initedCh = make(chan struct{})
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// Copy the channel since c.initCh can be set as nil after clock.RegisterPing.
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initCh := c.initCh
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clock.RegisterPing(c.ping)
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// Initialize oto.Player lazily to enable calling NewContext in an 'init' function.
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// Accessing oto.Player functions requires the environment to be already initialized,
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// but if Ebiten is used for a shared library, the timing when init functions are called
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// is unexpectable.
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// e.g. a variable for JVM on Android might not be set.
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<-initCh
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// This is a heuristic decision of audio buffer size.
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// On most desktops, 1/30[s] is enough but there are some known environment that is too short (e.g. Windows on Parallels).
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// On desktop browsers, 1/15[s] should work with any sample rate except for mobile browsers.
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// On mobiles, we don't have enough data. For iOS, 1/30[s] is too short and 1/20[s] seems fine. 1/15[s] is safer.
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bufferSize := c.sampleRate * channelNum * bytesPerSample / 15
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if web.IsAndroidChrome() {
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// On Android Chrome, it looks like 9600 * 4 is a sweet spot of the buffer size
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// regardless of the sample rate. This is about 1/5[s] for 48000[Hz].
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bufferSize = 9600 * channelNum * bytesPerSample
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}
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p, err := oto.NewPlayer(c.sampleRate, channelNum, bytesPerSample, bufferSize)
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if err != nil {
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audiobinding.SetError(err)
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return
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}
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defer p.Close()
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close(c.initedCh)
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for {
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c.m.Lock()
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if c.pingCount == 0 {
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c.m.Unlock()
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time.Sleep(10 * time.Millisecond)
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continue
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}
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c.pingCount--
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c.m.Unlock()
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c.frames++
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clock.ProceedPrimaryTimer()
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bytesPerFrame := c.sampleRate * bytesPerSample * channelNum / clock.FPS
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l := (c.frames * int64(bytesPerFrame)) - c.writtenBytes
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l &= mask
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c.writtenBytes += l
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buf := make([]uint8, l)
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if _, err := io.ReadFull(c.players, buf); err != nil {
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audiobinding.SetError(err)
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return
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}
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if _, err = p.Write(buf); err != nil {
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audiobinding.SetError(err)
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return
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}
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}
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}
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// Update is deprecated as of 1.6.0-alpha.
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//
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// As of 1.6.0-alpha, Update always returns nil and does nothing related to updating the state.
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// You don't have to call this function any longer.
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// The internal audio error is returned at ebiten.Run instead.
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func (c *Context) Update() error {
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return nil
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}
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// SampleRate returns the sample rate.
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func (c *Context) SampleRate() int {
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return c.sampleRate
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}
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// ReadSeekCloser is an io.ReadSeeker and io.Closer.
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type ReadSeekCloser interface {
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io.ReadSeeker
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io.Closer
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}
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type bytesReadSeekCloser struct {
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reader *bytes.Reader
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}
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func (b *bytesReadSeekCloser) Read(buf []uint8) (int, error) {
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return b.reader.Read(buf)
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}
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func (b *bytesReadSeekCloser) Seek(offset int64, whence int) (int64, error) {
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return b.reader.Seek(offset, whence)
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}
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func (b *bytesReadSeekCloser) Close() error {
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b.reader = nil
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return nil
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}
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// BytesReadSeekCloser creates ReadSeekCloser from bytes.
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func BytesReadSeekCloser(b []uint8) ReadSeekCloser {
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return &bytesReadSeekCloser{reader: bytes.NewReader(b)}
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}
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type readingResult struct {
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data []uint8
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err error
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}
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// Player is an audio player which has one stream.
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type Player struct {
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players *players
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src ReadSeekCloser
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sampleRate int
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buf []uint8
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pos int64
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volume float64
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seeking bool
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nextPos int64
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srcM sync.Mutex
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m sync.RWMutex
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}
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// NewPlayer creates a new player with the given stream.
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//
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// src's format must be linear PCM (16bits little endian, 2 channel stereo)
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// without a header (e.g. RIFF header).
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// The sample rate must be same as that of the audio context.
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//
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// Note that the given src can't be shared with other Player objects.
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//
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// NewPlayer tries to call Seek of src to get the current position.
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// NewPlayer returns error when the Seek returns error.
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func NewPlayer(context *Context, src ReadSeekCloser) (*Player, error) {
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if context.players.hasSource(src) {
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return nil, errors.New("audio: src cannot be shared with another Player")
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}
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p := &Player{
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players: context.players,
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src: src,
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sampleRate: context.sampleRate,
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buf: []uint8{},
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volume: 1,
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}
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// Get the current position of the source.
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pos, err := p.src.Seek(0, io.SeekCurrent)
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if err != nil {
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return nil, err
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}
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p.pos = pos
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runtime.SetFinalizer(p, (*Player).Close)
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return p, nil
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}
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// NewPlayerFromBytes creates a new player with the given bytes.
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//
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// As opposed to NewPlayer, you don't have to care if src is already used by another player or not.
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// src can be shared by multiple players.
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//
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// The format of src should be same as noted at NewPlayer.
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//
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// NewPlayerFromBytes's error is always nil as of 1.5.0-alpha.
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func NewPlayerFromBytes(context *Context, src []uint8) (*Player, error) {
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b := BytesReadSeekCloser(src)
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p, err := NewPlayer(context, b)
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if err != nil {
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// Errors should never happen.
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panic(err)
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}
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return p, nil
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}
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// Close closes the stream.
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//
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// When closing, the stream owned by the player will also be closed by calling its Close.
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// This means that the source stream passed via NewPlayer will also be closed.
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//
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// Close returns error when closing the source returns error.
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func (p *Player) Close() error {
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p.players.removePlayer(p)
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runtime.SetFinalizer(p, nil)
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p.srcM.Lock()
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err := p.src.Close()
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p.srcM.Unlock()
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return err
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}
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func (p *Player) readToBuffer(length int) (int, error) {
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b := make([]uint8, length)
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p.srcM.Lock()
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n, err := p.src.Read(b)
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p.srcM.Unlock()
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if err != nil {
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return 0, err
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}
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p.buf = append(p.buf, b[:n]...)
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return len(p.buf), nil
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}
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func (p *Player) bufferToInt16(lengthInBytes int) []int16 {
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r := make([]int16, lengthInBytes/2)
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// This function must be called on the same goruotine of readToBuffer.
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p.m.RLock()
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for i := 0; i < lengthInBytes/2; i++ {
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r[i] = int16(p.buf[2*i]) | (int16(p.buf[2*i+1]) << 8)
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r[i] = int16(float64(r[i]) * p.volume)
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}
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p.m.RUnlock()
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return r
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}
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func (p *Player) proceed(length int) {
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// This function must be called on the same goruotine of readToBuffer.
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p.m.Lock()
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p.buf = p.buf[length:]
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p.pos += int64(length)
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p.m.Unlock()
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}
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// Play plays the stream.
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//
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// Play always returns nil.
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func (p *Player) Play() error {
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p.players.addPlayer(p)
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return nil
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}
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// IsPlaying returns boolean indicating whether the player is playing.
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func (p *Player) IsPlaying() bool {
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return p.players.hasPlayer(p)
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}
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// Rewind rewinds the current position to the start.
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//
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// Rewind returns error when seeking the source stream returns error.
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func (p *Player) Rewind() error {
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return p.Seek(0)
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}
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// Seek seeks the position with the given offset.
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//
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// Seek returns error when seeking the source stream returns error.
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func (p *Player) Seek(offset time.Duration) error {
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o := int64(offset) * bytesPerSample * channelNum * int64(p.sampleRate) / int64(time.Second)
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o &= mask
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p.srcM.Lock()
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pos, err := p.src.Seek(o, io.SeekStart)
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p.srcM.Unlock()
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if err != nil {
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return err
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}
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p.m.Lock()
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p.seeking = true
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p.nextPos = pos
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p.m.Unlock()
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return nil
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}
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func (p *Player) resetBufferIfSeeking() {
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// This function must be called on the same goruotine of readToBuffer.
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p.m.Lock()
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if p.seeking {
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p.buf = []uint8{}
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p.pos = p.nextPos
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p.seeking = false
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}
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p.m.Unlock()
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}
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// Pause pauses the playing.
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//
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// Pause always returns nil.
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func (p *Player) Pause() error {
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p.players.removePlayer(p)
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return nil
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}
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// Current returns the current position.
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func (p *Player) Current() time.Duration {
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p.m.RLock()
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sample := p.pos / bytesPerSample / channelNum
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t := time.Duration(sample) * time.Second / time.Duration(p.sampleRate)
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p.m.RUnlock()
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return t
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}
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// Volume returns the current volume of this player [0-1].
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func (p *Player) Volume() float64 {
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p.m.RLock()
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v := p.volume
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p.m.RUnlock()
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return v
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}
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// SetVolume sets the volume of this player.
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// volume must be in between 0 and 1. This function panics otherwise.
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func (p *Player) SetVolume(volume float64) {
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p.m.Lock()
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// The condition must be true when volume is NaN.
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if !(0 <= volume && volume <= 1) {
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panic("audio: volume must be in between 0 and 1")
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}
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p.volume = volume
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p.m.Unlock()
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}
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