Added volume adjust sound behind the flag.

media/audio/sounds/wav_audio_handler.cc

Index: media/audio/sounds/wav_audio_handler.cc
diff --git a/media/audio/sounds/wav_audio_handler.cc b/media/audio/sounds/wav_audio_handler.cc
index 20eab8be4377f908451905f53014326ca1994c03..5479c1e2dfbe2fb843eeb846ac999fa4c77f1925 100644
--- a/media/audio/sounds/wav_audio_handler.cc
+++ b/media/audio/sounds/wav_audio_handler.cc
@@ -39,8 +39,12 @@ const size_t kBitsPerSampleOffset = 14;
 // Some constants for audio format.
 const int kAudioFormatPCM = 1;
 
+// The number of frames each OnMoreData() call will request.
+const int kDefaultFrameCount = 1024;
+
 // Reads an integer from |data| with |offset|.
-template<typename T> T ReadInt(const base::StringPiece& data, size_t offset) {
+template <typename T>
+T ReadInt(const base::StringPiece& data, size_t offset) {
   CHECK_LE(offset + sizeof(T), data.size());
   T result;
   memcpy(&result, data.data() + offset, sizeof(T));
@@ -54,11 +58,7 @@ template<typename T> T ReadInt(const base::StringPiece& data, size_t offset) {
 
 namespace media {
 
-WavAudioHandler::WavAudioHandler(const base::StringPiece& wav_data)
-    : num_channels_(0),
-      sample_rate_(0),
-      byte_rate_(0),
-      bits_per_sample_(0) {
+WavAudioHandler::WavAudioHandler(const base::StringPiece& wav_data) {
   CHECK_LE(kWavFileHeaderSize, wav_data.size()) << "wav data is too small";
   CHECK(wav_data.starts_with(kChunkId) &&
         memcmp(wav_data.data() + 8, kFormat, 4) == 0)
@@ -72,11 +72,15 @@ WavAudioHandler::WavAudioHandler(const base::StringPiece& wav_data)
     CHECK_LE(0, length) << "can't parse wav sub-chunk";
     offset += length;
   }
-}
 
-WavAudioHandler::~WavAudioHandler() {
+  const int64 size = data_.size();

[DaleCurtis, 2013/12/18 21:20:21]

If you set AudioBuffer::frames_per_buffer = data_.size() / (channels *
bytes_per_sample), the duration returned by AudioParameters::GetBufferDuration
should match this calculation.

[ygorshenin1, 2013/12/19 15:42:29]

frames_per_buffer is used in creation of AudioStreamProxy, and when the change
you proposed is applied, volume up/down sound becomes very harsh. I'd prefer to
use kDefaultFrameCount here.

[DaleCurtis, 2013/12/19 19:44:08]

Yes, you shouldn't use the AudioParameters directly from WavAudioHandler in
stream creation.  AudioStreamHandler should choose its own buffer size, it can
copy the rest of these parameters though.

[ygorshenin1, 2013/12/20 09:05:05]

Done.

+  const int64 rate = params_.GetBytesPerSecond();
+  if (rate)
+    duration_ = base::TimeDelta::FromMicroseconds(size * 1000000 / rate);
 }
 
+WavAudioHandler::~WavAudioHandler() {}
+
 bool WavAudioHandler::AtEnd(size_t cursor) const {
   return data_.size() <= cursor;
 }
@@ -86,18 +90,20 @@ bool WavAudioHandler::CopyTo(AudioBus* bus,
                              size_t* bytes_written) const {
   if (!bus)
     return false;
-  if (bus->channels() != num_channels_) {
-    LOG(ERROR) << "Number of channel mismatch.";
+  if (bus->channels() != params_.channels()) {
+    VLOG(1) << "Number of channel mismatch.";
     return false;
   }
   if (AtEnd(cursor)) {
     bus->Zero();
     return true;
   }
-  const int remaining_frames = (data_.size() - cursor) / bytes_per_frame_;
+  const int remaining_frames =
+      (data_.size() - cursor) / params_.GetBytesPerFrame();
   const int frames = std::min(bus->frames(), remaining_frames);
-  bus->FromInterleaved(data_.data() + cursor, frames, bytes_per_sample_);
-  *bytes_written = frames * bytes_per_frame_;
+  bus->FromInterleaved(data_.data() + cursor, frames,
+                       params_.bits_per_sample() >> 3);

[DaleCurtis, 2013/12/18 21:20:21]

Just use * 8, the compiler will optimize as necessary.

[ygorshenin1, 2013/12/19 15:42:29]

Done.

+  *bytes_written = frames * params_.GetBytesPerFrame();
   bus->ZeroFramesPartial(frames, bus->frames() - frames);
   return true;
 }
@@ -124,12 +130,14 @@ bool WavAudioHandler::ParseFmtChunk(const base::StringPiece& data) {
     return false;
   }
   DCHECK_EQ(ReadInt<uint16>(data, kAudioFormatOffset), kAudioFormatPCM);
-  num_channels_ = ReadInt<uint16>(data, kChannelOffset);
-  sample_rate_ = ReadInt<uint32>(data, kSampleRateOffset);
-  byte_rate_ = ReadInt<uint32>(data, kByteRateOffset);
-  bits_per_sample_ = ReadInt<uint16>(data, kBitsPerSampleOffset);
-  bytes_per_sample_ = bits_per_sample_ >> 3;
-  bytes_per_frame_ = num_channels_ * bytes_per_sample_;
+  int num_channels = ReadInt<uint16>(data, kChannelOffset);
+  int sample_rate = ReadInt<uint32>(data, kSampleRateOffset);
+  int bits_per_sample = ReadInt<uint16>(data, kBitsPerSampleOffset);
+  params_ = AudioParameters(AudioParameters::AUDIO_PCM_LOW_LATENCY,
+                            GuessChannelLayout(num_channels),
+                            sample_rate,
+                            bits_per_sample,
+                            kDefaultFrameCount);
   return true;
 }