Always use the source channel layout with AudioBufferConverter.

media/filters/audio_renderer_impl.cc

 // 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.
 
 #include "media/filters/audio_renderer_impl.h"
 
 #include <math.h>
 
 #include <algorithm>
 
@@ skipping 265 matching lines @@
         ChannelLayoutToChannelCount(
             stream->audio_decoder_config().channel_layout()),
         0,
         stream->audio_decoder_config().samples_per_second(),
         stream->audio_decoder_config().bits_per_channel(),
         buffer_size);
     buffer_converter_.reset();
   } else {
     // TODO(rileya): Support hardware config changes
     const AudioParameters& hw_params = hardware_config_->GetOutputConfig();
-    audio_parameters_.Reset(hw_params.format(),
-                            hw_params.channel_layout(),
-                            hw_params.channels(),
-                            hw_params.input_channels(),
-                            hw_params.sample_rate(),
-                            hw_params.bits_per_sample(),
-                            hardware_config_->GetHighLatencyBufferSize());
+    audio_parameters_.Reset(
+        hw_params.format(),
+        // Always use the source's channel layout and channel count to avoid
+        // premature downmixing (http://crbug.com/379288), platform specific
+        // issues around channel layouts (http://crbug.com/266674), and
+        // unnecessary upmixing overhead.
+        stream->audio_decoder_config().channel_layout(),
+        ChannelLayoutToChannelCount(
+            stream->audio_decoder_config().channel_layout()),
+        hw_params.input_channels(),
+        hw_params.sample_rate(),
+        hw_params.bits_per_sample(),
+        hardware_config_->GetHighLatencyBufferSize());
   }
 
   audio_clock_.reset(new AudioClock(audio_parameters_.sample_rate()));
 
   audio_buffer_stream_.Initialize(
       stream,
       false,
       statistics_cb,
       base::Bind(&AudioRendererImpl::OnAudioBufferStreamInitialized,
                  weak_factory_.GetWeakPtr()));
@@ skipping 453 matching lines @@
   DCHECK(expecting_config_changes_);
   buffer_converter_->ResetTimestampState();
   // Drain flushed buffers from the converter so the AudioSplicer receives all
   // data ahead of any OnNewSpliceBuffer() calls.  Since discontinuities should
   // only appear after config changes, AddInput() should never fail here.
   while (buffer_converter_->HasNextBuffer())
     CHECK(splicer_->AddInput(buffer_converter_->GetNextBuffer()));
 }
 
 }  // namespace media