Revert 258215 "Remove muting for extreme playbackRates."

media/filters/audio_renderer_algorithm.h

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 // AudioRendererAlgorithm buffers and transforms audio data. The owner of
 // this object provides audio data to the object through EnqueueBuffer() and
 // requests data from the buffer via FillBuffer(). The owner also sets the
 // playback rate, and the AudioRendererAlgorithm will stretch or compress the
 // buffered audio as necessary to match the playback rate when fulfilling
 // FillBuffer() requests.
 //
 // This class is *not* thread-safe. Calls to enqueue and retrieve data must be
 // locked if called from multiple threads.
 //
 // AudioRendererAlgorithm uses the Waveform Similarity Overlap and Add (WSOLA)
 // algorithm to stretch or compress audio data to meet playback speeds less than
 // or greater than the natural playback of the audio stream. The algorithm
 // preserves local properties of the audio, therefore, pitch and harmonics are
 // 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_
 
 #include "base/memory/ref_counted.h"
 #include "base/memory/scoped_ptr.h"
 #include "media/audio/audio_parameters.h"
 #include "media/base/audio_buffer.h"
 #include "media/base/audio_buffer_queue.h"
 
@@ skipping 43 matching lines @@
   void IncreaseQueueCapacity();
 
   // Returns the number of frames left in |audio_buffer_|, which may be larger
   // than QueueCapacity() in the event that EnqueueBuffer() delivered more data
   // than |audio_buffer_| was intending to hold.
   int frames_buffered() { return audio_buffer_.frames(); }
 
   // 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
   // there is enough data to perform a search, i.e. |search_block_| and
   // |target_block_| can be extracted from the available frames.
   void GetOptimalBlock();
 
   // Read a maximum of |requested_frames| frames from |wsola_output_|. Returns
   // number of frames actually read.
   int WriteCompletedFramesTo(
@@ skipping 33 matching lines @@
 
   // Sample rate of audio stream.
   int samples_per_second_;
 
   // Used by algorithm to scale output.
   float playback_rate_;
 
   // 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_;
 
   // Book keeping of the current time of generated audio, in frames. This
   // should be appropriately updated when out samples are generated, regardless
   // of whether we push samples out when FillBuffer() is called or we store
   // audio in |wsola_output_| for the subsequent calls to FillBuffer().
   // Furthermore, if samples from |audio_buffer_| are evicted then this
   // member variable should be updated based on |playback_rate_|.
   // Note that this member should be updated ONLY by calling UpdateOutputTime(),
@@ skipping 53 matching lines @@
   // searched for a block (|optimal_block_|) that is most similar to
   // |target_block_|.
   scoped_ptr<AudioBus> target_block_;
 
   DISALLOW_COPY_AND_ASSIGN(AudioRendererAlgorithm);
 };
 
 }  // namespace media
 
 #endif  // MEDIA_FILTERS_AUDIO_RENDERER_ALGORITHM_H_