Fixes bug in FFmpegAudioDecoder where we weren't setting the usable data size.

media/filters/ffmpeg_audio_decoder.cc

 // Copyright (c) 2009 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.
 
 #include "media/base/data_buffer.h"
 #include "media/filters/ffmpeg_audio_decoder.h"
 #include "media/filters/ffmpeg_common.h"
 #include "media/filters/ffmpeg_demuxer.h"
 
 namespace media {
@@ skipping 90 matching lines @@
   if (result < 0 ||
       output_buffer_size < 0 ||
       static_cast<size_t>(output_buffer_size) > kOutputBufferSize) {
     host()->Error(PIPELINE_ERROR_DECODE);
     return;
   }
 
   // If we have decoded something, enqueue the result.
   if (output_buffer_size) {
     DataBuffer* result_buffer = new DataBuffer(output_buffer_size);
+    result_buffer->SetDataSize(output_buffer_size);
     uint8* data = result_buffer->GetWritableData();
     memcpy(data, output_buffer, output_buffer_size);
 
     // Determine the duration if the demuxer couldn't figure it out, otherwise
     // copy it over.
     if (input->GetDuration().InMicroseconds() == 0) {
       result_buffer->SetDuration(CalculateDuration(output_buffer_size));
     } else {
       result_buffer->SetDuration(input->GetDuration());
     }
@@ skipping 20 matching lines @@
 base::TimeDelta FFmpegAudioDecoder::CalculateDuration(size_t size) {
   int64 denominator = codec_context_->channels *
       av_get_bits_per_sample_format(codec_context_->sample_fmt) / 8 *
       codec_context_->sample_rate;
   double microseconds = size /
       (denominator / static_cast<double>(base::Time::kMicrosecondsPerSecond));
   return base::TimeDelta::FromMicroseconds(static_cast<int64>(microseconds));
 }
 
 }  // namespace