Fix accessing uninitialized variables when not processing a reverse stream.

webrtc/modules/audio_processing/test/audioproc_float.cc

 /*
  *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 150 matching lines @@
   TickTime processing_start_time;
   TickInterval accumulated_time;
   int num_chunks = 0;
 
   const StreamConfig input_config = {
       in_file.sample_rate(), in_buf.num_channels(),
   };
   const StreamConfig output_config = {
       out_file.sample_rate(), out_buf.num_channels(),
   };
-  const StreamConfig reverse_input_config = {
-      in_rev_file->sample_rate(), in_rev_file->num_channels(),
-  };
-  const StreamConfig reverse_output_config = {
-      out_rev_file->sample_rate(), out_rev_file->num_channels(),
-  };
+
+  StreamConfig reverse_input_config = {};
+  StreamConfig reverse_output_config = {};
+  if (process_reverse) {
+    StreamConfig reverse_input_config = {in_rev_file->sample_rate(),

[Andrew MacDonald, 2015/08/21 17:31:33]

This redeclaration is a bug of course; fixed here:
https://codereview.webrtc.org/1310653002/

+                                         in_rev_file->num_channels()};
+    StreamConfig reverse_output_config = {out_rev_file->sample_rate(),
+                                          out_rev_file->num_channels()};
+  }
   while (in_file.ReadSamples(in_interleaved.size(),
                              &in_interleaved[0]) == in_interleaved.size()) {
     // Have logs display the file time rather than wallclock time.
     trace_to_stderr.SetTimeSeconds(num_chunks * 1.f / kChunksPerSecond);
     FloatS16ToFloat(&in_interleaved[0], in_interleaved.size(),
                     &in_interleaved[0]);
     Deinterleave(&in_interleaved[0], in_buf.num_frames(),
                  in_buf.num_channels(), in_buf.channels());
     if (process_reverse) {
       in_rev_file->ReadSamples(in_rev_interleaved.size(),
@@ skipping 41 matching lines @@
            execution_time_ms * 1.f / num_chunks);
   }
   return 0;
 }
 
 }  // namespace webrtc
 
 int main(int argc, char* argv[]) {
   return webrtc::main(argc, argv);
 }