Use AudioClock to determine when audio playback has ended.

media/filters/audio_renderer_impl.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.
 
 // Audio rendering unit utilizing an AudioRendererSink to output data.
 //
 // This class lives inside three threads during it's lifetime, namely:
 // 1. Render thread
 //    Where the object is created.
 // 2. Media thread (provided via constructor)
@@ skipping 109 matching lines @@
                          const scoped_refptr<AudioBuffer>& buffer);
 
   // Handles buffers that come out of |splicer_|.
   // Returns true if more buffers are needed.
   bool HandleSplicerBuffer_Locked(const scoped_refptr<AudioBuffer>& buffer);
 
   // Helper functions for AudioDecoder::Status values passed to
   // DecodedAudioReady().
   void HandleAbortedReadOrDecodeError(bool is_decode_error);
 
-  // Estimate earliest time when current buffer can stop playing.
-  void UpdateEarliestEndTime_Locked(int frames_filled,
-                                    const base::TimeDelta& playback_delay,
-                                    const base::TimeTicks& time_now);
-
   void StartRendering_Locked();
   void StopRendering_Locked();
 
   // AudioRendererSink::RenderCallback implementation.
   //
   // NOTE: These are called on the audio callback thread!
   //
   // Render() fills the given buffer with audio data by delegating to its
   // |algorithm_|. Render() also takes care of updating the clock.
   // Returns the number of frames copied into |audio_bus|, which may be less
@@ skipping 105 matching lines @@
   bool pending_read_;
 
   // Keeps track of whether we received and rendered the end of stream buffer.
   bool received_end_of_stream_;
   bool rendered_end_of_stream_;
 
   scoped_ptr<AudioClock> audio_clock_;
 
   base::TimeDelta start_timestamp_;
 
-  // We're supposed to know amount of audio data OS or hardware buffered, but
-  // that is not always so -- on my Linux box
-  // AudioBuffersState::hardware_delay_bytes never reaches 0.
-  //
-  // As a result we cannot use it to find when stream ends. If we just ignore
-  // buffered data we will notify host that stream ended before it is actually
-  // did so, I've seen it done ~140ms too early when playing ~150ms file.
-  //
-  // Instead of trying to invent OS-specific solution for each and every OS we
-  // are supporting, use simple workaround: every time we fill the buffer we
-  // remember when it should stop playing, and do not assume that buffer is
-  // empty till that time. Workaround is not bulletproof, as we don't exactly
-  // know when that particular data would start playing, but it is much better
-  // than nothing.
-  base::TimeTicks earliest_end_time_;
-  size_t total_frames_filled_;

[scherkus (not reviewing), 2014/07/10 00:33:44]

apparently this variable was completely unused

[DaleCurtis, 2014/07/10 01:44:05]

Haha, yeah, I've removed it a couple times locally, but I guess I forgot to
upload those patch sets. Thanks!

-
   // End variables which must be accessed under |lock_|. ----------------------
 
   // NOTE: Weak pointers must be invalidated before all other member variables.
   base::WeakPtrFactory<AudioRendererImpl> weak_factory_;
 
   DISALLOW_COPY_AND_ASSIGN(AudioRendererImpl);
 };
 
 }  // namespace media
 
 #endif  // MEDIA_FILTERS_AUDIO_RENDERER_IMPL_H_