Revert 258215 "Remove muting for extreme playbackRates."

It exposed a bug in the WSOLA algorithm that causes it to incorrectly
trigger underflows.

> Remove muting for extreme playbackRates.
> 
> Audio was muted below 0.5x and above 4x as the quality degraded
> significantly under the crossfade algorithm. The quality is now much
> better under the WSLOA algorithm (r220343).
> 
> BUG=289354
> R=scherkus@chromium.org
> R=dalecurtis@chromium.org
> 
> Review URL: https://codereview.chromium.org/205093002

BUG=368083
TBR=sandersd@chromium.org

Review URL: https://codereview.chromium.org/312963003

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@274701 0039d316-1c4b-4281-b951-d872f2087c98

media/filters/audio_renderer_algorithm.cc

@@ -46,6 +46,12 @@ namespace media {
 //    |search_block_index_| = |search_block_center_offset_| -
 //        |search_block_center_offset_|.
 
+// Max/min supported playback rates for fast/slow audio. Audio outside of these
+// ranges are muted.
+// Audio at these speeds would sound better under a frequency domain algorithm.
+static const float kMinPlaybackRate = 0.5f;
+static const float kMaxPlaybackRate = 4.0f;
+
 // Overlap-and-add window size in milliseconds.
 static const int kOlaWindowSizeMs = 20;
 
@@ -70,6 +76,8 @@ AudioRendererAlgorithm::AudioRendererAlgorithm()
     : channels_(0),
       samples_per_second_(0),
       playback_rate_(0),
+      muted_(false),
+      muted_partial_frame_(0),
       capacity_(kStartingBufferSizeInFrames),
       output_time_(0.0),
       search_block_center_offset_(0),
@@ -143,6 +151,31 @@ int AudioRendererAlgorithm::FillBuffer(AudioBus* dest, int requested_frames) {
 
   DCHECK_EQ(channels_, dest->channels());
 
+  // Optimize the |muted_| case to issue a single clear instead of performing
+  // the full crossfade and clearing each crossfaded frame.
+  if (muted_) {
+    int frames_to_render =
+        std::min(static_cast<int>(audio_buffer_.frames() / playback_rate_),
+                 requested_frames);
+
+    // Compute accurate number of frames to actually skip in the source data.
+    // Includes the leftover partial frame from last request. However, we can
+    // only skip over complete frames, so a partial frame may remain for next
+    // time.
+    muted_partial_frame_ += frames_to_render * playback_rate_;
+    int seek_frames = static_cast<int>(muted_partial_frame_);
+    dest->ZeroFrames(frames_to_render);
+    audio_buffer_.SeekFrames(seek_frames);
+
+    // Determine the partial frame that remains to be skipped for next call. If
+    // the user switches back to playing, it may be off time by this partial
+    // frame, which would be undetectable. If they subsequently switch to
+    // another playback rate that mutes, the code will attempt to line up the
+    // frames again.
+    muted_partial_frame_ -= seek_frames;
+    return frames_to_render;
+  }
+
   int slower_step = ceil(ola_window_size_ * playback_rate_);
   int faster_step = ceil(ola_window_size_ / playback_rate_);
 
@@ -167,6 +200,8 @@ int AudioRendererAlgorithm::FillBuffer(AudioBus* dest, int requested_frames) {
 void AudioRendererAlgorithm::SetPlaybackRate(float new_rate) {
   DCHECK_GE(new_rate, 0);
   playback_rate_ = new_rate;
+  muted_ =
+      playback_rate_ < kMinPlaybackRate || playback_rate_ > kMaxPlaybackRate;
 }
 
 void AudioRendererAlgorithm::FlushBuffers() {

media/filters/audio_renderer_algorithm.h

@@ -19,6 +19,8 @@
 // are preserved. See audio_renderer_algorith.cc for a more elaborate
 // description of the algorithm.
 //
+// Audio at very low or very high playback rates are muted to preserve quality.
+//
 
 #ifndef MEDIA_FILTERS_AUDIO_RENDERER_ALGORITHM_H_
 #define MEDIA_FILTERS_AUDIO_RENDERER_ALGORITHM_H_
@@ -82,6 +84,9 @@ class MEDIA_EXPORT AudioRendererAlgorithm {
   // Returns the samples per second for this audio stream.
   int samples_per_second() { return samples_per_second_; }
 
+  // Is the sound currently muted?
+  bool is_muted() { return muted_; }
+
  private:
   // Within |search_block_|, find the block of data that is most similar to
   // |target_block_|, and write it in |optimal_block_|. This method assumes that
@@ -135,6 +140,12 @@ class MEDIA_EXPORT AudioRendererAlgorithm {
   // Buffered audio data.
   AudioBufferQueue audio_buffer_;
 
+  // True if the audio should be muted.
+  bool muted_;
+
+  // If muted, keep track of partial frames that should have been skipped over.
+  double muted_partial_frame_;
+
   // How many frames to have in the queue before we report the queue is full.
   int capacity_;
 

media/filters/audio_renderer_algorithm_unittest.cc

@@ -151,7 +151,7 @@ class AudioRendererAlgorithmTest : public testing::Test {
       bool all_zero = true;
       for (int i = 0; i < frames_written && all_zero; ++i)
         all_zero = audio_data->channel(ch)[i] == 0.0f;
-      ASSERT_FALSE(all_zero) << " for channel " << ch;
+      ASSERT_EQ(algorithm_.is_muted(), all_zero) << " for channel " << ch;
     }
   }