audio: Auto resampling (#301)

audio/internal/resampling/resampling.go

@@ -0,0 +1,172 @@
+// Copyright 2017 The Ebiten 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 resampling
+
+import (
+	"io"
+	"math"
+
+	"github.com/hajimehoshi/ebiten/audio"
+)
+
+func sinc(x float64) float64 {
+	if x == 0 {
+		return 1
+	}
+	return math.Sin(x) / x
+}
+
+type Stream struct {
+	source    audio.ReadSeekCloser
+	size      int64
+	from      int
+	to        int
+	pos       int64
+	srcPos    int64
+	srcCacheL []float64
+	srcCacheR []float64
+}
+
+func NewStream(source audio.ReadSeekCloser, size int64, from, to int) *Stream {
+	s := &Stream{
+		source: source,
+		size:   size,
+		from:   from,
+		to:     to,
+	}
+	s.srcCacheL = make([]float64, s.size/4)
+	s.srcCacheR = make([]float64, s.size/4)
+	return s
+}
+
+func (s *Stream) Size() int64 {
+	return int64(float64(s.size) * float64(s.to) / float64(s.from))
+}
+
+func (s *Stream) src(i int) (float64, float64, error) {
+	// Use int here since int64 is very slow on browsers.
+	// TODO: Resampling is too heavy on browsers. How about using OfflineAudioContext?
+	if i < 0 {
+		return 0, 0, nil
+	}
+	if len(s.srcCacheL) <= i {
+		return 0, 0, nil
+	}
+	pos := int(s.srcPos) / 4
+	if pos <= i {
+		buf := make([]uint8, 4096)
+		n, err := s.source.Read(buf)
+		if err != nil && err != io.EOF {
+			return 0, 0, err
+		}
+		n = n / 4 * 4
+		buf = buf[:n]
+		for i := 0; i < len(buf)/4; i++ {
+			srcL := float64(int16(buf[4*i])|(int16(buf[4*i+1])<<8)) / (1<<15 - 1)
+			srcR := float64(int16(buf[4*i+2])|(int16(buf[4*i+3])<<8)) / (1<<15 - 1)
+			s.srcCacheL[pos+i] = srcL
+			s.srcCacheR[pos+i] = srcR
+		}
+		s.srcPos += int64(n)
+	}
+	return s.srcCacheL[i], s.srcCacheR[i], nil
+}
+
+func (s *Stream) at(t int64) (float64, float64, error) {
+	const windowSize = 8
+	tInSrc := float64(t) * float64(s.from) / float64(s.to)
+	startN := int64(tInSrc) - windowSize
+	if startN < 0 {
+		startN = 0
+	}
+	if s.size/4 < startN {
+		startN = s.size / 4
+	}
+	endN := int64(tInSrc) + windowSize + 1
+	if s.size/4 < endN {
+		endN = s.size / 4
+	}
+	l := 0.0
+	r := 0.0
+	for n := startN; n < endN; n++ {
+		srcL, srcR, err := s.src(int(n))
+		if err != nil {
+			return 0, 0, err
+		}
+		w := 0.5 + 0.5*math.Cos(2*math.Pi*(tInSrc-float64(n))/(windowSize*2+1))
+		s := sinc(math.Pi*(tInSrc-float64(n))) * w
+		l += srcL * s
+		r += srcR * s
+	}
+	if l < -1 {
+		l = -1
+	}
+	if l > 1 {
+		l = 1
+	}
+	if r < -1 {
+		r = -1
+	}
+	if r > 1 {
+		r = 1
+	}
+	return l, r, nil
+}
+
+func (s *Stream) Read(b []uint8) (int, error) {
+	if s.pos == s.Size() {
+		return 0, io.EOF
+	}
+	n := len(b) / 4 * 4
+	if s.Size()-s.pos <= int64(n) {
+		n = int(s.Size() - s.pos)
+	}
+	for i := 0; i < n/4; i++ {
+		l, r, err := s.at(s.pos/4 + int64(i))
+		if err != nil {
+			return 0, err
+		}
+		l16 := int16(l * (1<<15 - 1))
+		r16 := int16(r * (1<<15 - 1))
+		b[4*i] = uint8(l16)
+		b[4*i+1] = uint8(l16 >> 8)
+		b[4*i+2] = uint8(r16)
+		b[4*i+3] = uint8(r16 >> 8)
+	}
+	s.pos += int64(n)
+	return n, nil
+}
+
+func (s *Stream) Seek(offset int64, whence int) (int64, error) {
+	switch whence {
+	case io.SeekStart:
+		s.pos = offset
+	case io.SeekCurrent:
+		s.pos += offset
+	case io.SeekEnd:
+		s.pos += s.Size() + offset
+	}
+	if s.pos < 0 {
+		s.pos = 0
+	}
+	if s.Size() <= s.pos {
+		s.pos = s.Size()
+	}
+	return s.pos, nil
+}
+
+func (s *Stream) Close() error {
+	return s.source.Close()
+}

audio/vorbis/vorbis.go

@@ -21,12 +21,18 @@ import (
 	"runtime"
 
 	"github.com/hajimehoshi/ebiten/audio"
+	"github.com/hajimehoshi/ebiten/audio/internal/resampling"
 	"github.com/jfreymuth/oggvorbis"
 )
 
+type readSeekCloseSizer interface {
+	audio.ReadSeekCloser
+	Size() int64
+}
+
 // Stream is a decoded audio stream.
 type Stream struct {
-	decoded *decoded
+	decoded readSeekCloseSizer
 }
 
 // Read is implementation of io.Reader's Read.
@@ -172,7 +178,7 @@ func decode(in audio.ReadSeekCloser) (*decoded, int, int, error) {
 
 // Decode decodes Ogg/Vorbis data to playable stream.
 //
-// The sample rate must be same as that of audio context.
+// Sample rate is automatically adjusted to fit with the audio context.
 func Decode(context *audio.Context, src audio.ReadSeekCloser) (*Stream, error) {
 	decoded, channelNum, sampleRate, err := decode(src)
 	if err != nil {
@@ -183,10 +189,8 @@ func Decode(context *audio.Context, src audio.ReadSeekCloser) (*Stream, error) {
 		return nil, fmt.Errorf("vorbis: number of channels must be 2")
 	}
 	if sampleRate != context.SampleRate() {
-		return nil, fmt.Errorf("vorbis: sample rate must be %d but %d", context.SampleRate(), sampleRate)
+		s := resampling.NewStream(decoded, decoded.Size(), sampleRate, context.SampleRate())
+		return &Stream{s}, nil
 	}
-	s := &Stream{
+	return &Stream{decoded}, nil
-		decoded: decoded,
-	}
-	return s, nil
 }

audio/wav/decode.go

@@ -21,10 +21,42 @@ import (
 	"io"
 
 	"github.com/hajimehoshi/ebiten/audio"
+	"github.com/hajimehoshi/ebiten/audio/internal/resampling"
 )
 
+type readSeekCloseSizer interface {
+	audio.ReadSeekCloser
+	Size() int64
+}
+
 // Stream is a decoded audio stream.
 type Stream struct {
+	inner readSeekCloseSizer
+}
+
+// Read is implementation of io.Reader's Read.
+func (s *Stream) Read(p []byte) (int, error) {
+	return s.inner.Read(p)
+}
+
+// Seek is implementation of io.Seeker's Seek.
+//
+// Note that Seek can take long since decoding is a relatively heavy task.
+func (s *Stream) Seek(offset int64, whence int) (int64, error) {
+	return s.inner.Seek(offset, whence)
+}
+
+// Read is implementation of io.Closer's Close.
+func (s *Stream) Close() error {
+	return s.inner.Close()
+}
+
+// Size returns the size of decoded stream in bytes.
+func (s *Stream) Size() int64 {
+	return s.inner.Size()
+}
+
+type stream struct {
 	src        audio.ReadSeekCloser
 	headerSize int64
 	dataSize   int64
@@ -32,7 +64,7 @@ type Stream struct {
 }
 
 // Read is implementation of io.Reader's Read.
-func (s *Stream) Read(p []byte) (int, error) {
+func (s *stream) Read(p []byte) (int, error) {
 	if s.remaining <= 0 {
 		return 0, io.EOF
 	}
@@ -45,7 +77,7 @@ func (s *Stream) Read(p []byte) (int, error) {
 }
 
 // Seek is implementation of io.Seeker's Seek.
-func (s *Stream) Seek(offset int64, whence int) (int64, error) {
+func (s *stream) Seek(offset int64, whence int) (int64, error) {
 	if whence == io.SeekStart {
 		offset += s.headerSize
 	}
@@ -66,19 +98,20 @@ func (s *Stream) Seek(offset int64, whence int) (int64, error) {
 }
 
 // Read is implementation of io.Closer's Close.
-func (s *Stream) Close() error {
+func (s *stream) Close() error {
 	return s.src.Close()
 }
 
 // Size returns the size of decoded stream in bytes.
-func (s *Stream) Size() int64 {
+func (s *stream) Size() int64 {
 	return s.dataSize
 }
 
 // Decode decodes WAV (RIFF) data to playable stream.
 //
 // The format must be 2 channels, 16bit little endian PCM.
-// The sample rate must be same as that of audio context.
+//
+// Sample rate is automatically adjusted to fit with the audio context.
 func Decode(context *audio.Context, src audio.ReadSeekCloser) (*Stream, error) {
 	buf := make([]byte, 12)
 	n, err := io.ReadFull(src, buf)
@@ -98,6 +131,8 @@ func Decode(context *audio.Context, src audio.ReadSeekCloser) (*Stream, error) {
 	// Read chunks
 	dataSize := int64(0)
 	headerSize := int64(0)
+	sampleRateFrom := 0
+	sampleRateTo := 0
 chunks:
 	for {
 		buf := make([]byte, 8)
@@ -139,7 +174,8 @@ chunks:
 			}
 			sampleRate := int64(buf[4]) | int64(buf[5])<<8 | int64(buf[6])<<16 | int64(buf[7])<<24
 			if int64(context.SampleRate()) != sampleRate {
-				return nil, fmt.Errorf("wav: sample rate must be %d but %d", context.SampleRate(), sampleRate)
+				sampleRateFrom = int(sampleRate)
+				sampleRateTo = context.SampleRate()
 			}
 			headerSize += size
 		case bytes.Equal(buf[0:4], []byte("data")):
@@ -157,11 +193,15 @@ chunks:
 			headerSize += size
 		}
 	}
-	s := &Stream{
+	s := &stream{
 		src:        src,
 		headerSize: headerSize,
 		dataSize:   dataSize,
 		remaining:  dataSize,
 	}
-	return s, nil
+	if sampleRateFrom != sampleRateTo {
+		fixed := resampling.NewStream(s, s.dataSize, sampleRateFrom, sampleRateTo)
+		return &Stream{fixed}, nil
+	}
+	return &Stream{s}, nil
 }

examples/audio/main.go

@@ -268,7 +268,9 @@ func main() {
 	if err != nil {
 		log.Fatal(err)
 	}
-	const sampleRate = 44100
+	// This sample rate doesn't match with wav/ogg's sample rate,
+	// but decoders adjust them.
+	const sampleRate = 48000
 	const bytesPerSample = 4 // TODO: This should be defined in audio package
 	audioContext, err = audio.NewContext(sampleRate)
 	if err != nil {