Remove AudioBuffer::set_duration(), instead base on frames.

media/base/audio_buffer.h

 // Copyright 2013 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.
 
 #ifndef MEDIA_BASE_AUDIO_BUFFER_H_
 #define MEDIA_BASE_AUDIO_BUFFER_H_
 
 #include <vector>
 
 #include "base/memory/aligned_memory.h"
@@ skipping 17 matching lines @@
   // (which may be 0, 16, or 32, depending on the processor). Selecting 32 in
   // order to work on all processors.
   enum { kChannelAlignment = 32 };
 
   // Create an AudioBuffer whose channel data is copied from |data|. For
   // interleaved data, only the first buffer is used. For planar data, the
   // number of buffers must be equal to |channel_count|. |frame_count| is the
   // number of frames in each buffer. |data| must not be null and |frame_count|
   // must be >= 0.
   //
   // TODO(jrummell): Compute duration rather than pass it in.

[wolenetz, 2014/05/01 20:18:24]

drive-by nit: remove this comment now?

[jrummell, 2014/05/01 20:26:26]

nit: Can you remove this comment since it's done now :)

[DaleCurtis, 2014/05/01 20:33:18]

Done.

   static scoped_refptr<AudioBuffer> CopyFrom(SampleFormat sample_format,
                                              ChannelLayout channel_layout,
                                              int channel_count,
                                              int sample_rate,
                                              int frame_count,
                                              const uint8* const* data,
-                                             const base::TimeDelta timestamp,
-                                             const base::TimeDelta duration);
+                                             const base::TimeDelta timestamp);
 
   // Create an AudioBuffer with |frame_count| frames. Buffer is allocated, but
   // not initialized. Timestamp and duration are set to kNoTimestamp().
   static scoped_refptr<AudioBuffer> CreateBuffer(SampleFormat sample_format,
                                                  ChannelLayout channel_layout,
                                                  int channel_count,
                                                  int sample_rate,
                                                  int frame_count);
 
   // Create an empty AudioBuffer with |frame_count| frames.
   static scoped_refptr<AudioBuffer> CreateEmptyBuffer(
       ChannelLayout channel_layout,
       int channel_count,
       int sample_rate,
       int frame_count,
-      const base::TimeDelta timestamp,
-      const base::TimeDelta duration);
+      const base::TimeDelta timestamp);
 
   // Create a AudioBuffer indicating we've reached end of stream.
   // Calling any method other than end_of_stream() on the resulting buffer
   // is disallowed.
   static scoped_refptr<AudioBuffer> CreateEOSBuffer();
 
   // Copy frames into |dest|. |frames_to_copy| is the number of frames to copy.
   // |source_frame_offset| specifies how many frames in the buffer to skip
   // first. |dest_frame_offset| is the frame offset in |dest|. The frames are
   // converted from their source format into planar float32 data (which is all
@@ skipping 21 matching lines @@
 
   // Return the sample rate.
   int sample_rate() const { return sample_rate_; }
 
   // Return the channel layout.
   ChannelLayout channel_layout() const { return channel_layout_; }
 
   base::TimeDelta timestamp() const { return timestamp_; }
   base::TimeDelta duration() const { return duration_; }
   void set_timestamp(base::TimeDelta timestamp) { timestamp_ = timestamp; }
-  void set_duration(base::TimeDelta duration) { duration_ = duration; }
 
   // If there's no data in this buffer, it represents end of stream.
   bool end_of_stream() const { return end_of_stream_; }
 
   // Access to the raw buffer for ffmpeg to write directly to. Data for planar
   // data is grouped by channel. There is only 1 entry for interleaved formats.
   const std::vector<uint8*>& channel_data() const { return channel_data_; }
 
  private:
   friend class base::RefCountedThreadSafe<AudioBuffer>;
 
   // Allocates aligned contiguous buffer to hold all channel data (1 block for
   // interleaved data, |channel_count| blocks for planar data), copies
   // [data,data+data_size) to the allocated buffer(s). If |data| is null, no
   // data is copied. If |create_buffer| is false, no data buffer is created (or
   // copied to).
   AudioBuffer(SampleFormat sample_format,
               ChannelLayout channel_layout,
               int channel_count,
               int sample_rate,
               int frame_count,
               bool create_buffer,
               const uint8* const* data,
-              const base::TimeDelta timestamp,
-              const base::TimeDelta duration);
+              const base::TimeDelta timestamp);
 
   virtual ~AudioBuffer();
 
   const SampleFormat sample_format_;
   const ChannelLayout channel_layout_;
   const int channel_count_;
   const int sample_rate_;
   int adjusted_frame_count_;
   int trim_start_;
   const bool end_of_stream_;
   base::TimeDelta timestamp_;
   base::TimeDelta duration_;
 
   // Contiguous block of channel data.
   scoped_ptr<uint8, base::AlignedFreeDeleter> data_;
 
   // For planar data, points to each channels data.
   std::vector<uint8*> channel_data_;
 
   DISALLOW_IMPLICIT_CONSTRUCTORS(AudioBuffer);
 };
 
 }  // namespace media
 
 #endif  // MEDIA_BASE_AUDIO_BUFFER_H_