Add a base class to Wav{Reader,Writer} to access shared parameters.

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 45 matching lines @@
 
 const int kChunksPerSecond = 100;
 const char kUsage[] =
     "Command-line tool to run audio processing on WAV files. Accepts either\n"
     "an input capture WAV file or protobuf debug dump and writes to an output\n"
     "WAV file.\n"
     "\n"
     "All components are disabled by default. If any bi-directional components\n"
     "are enabled, only debug dump files are permitted.";
 
+// Returns a StreamConfig corresponding to wav_file if it's non-nullptr.
+// Otherwise returns a default initialized StreamConfig.
+StreamConfig MakeStreamConfig(const WavFile* wav_file) {
+  if (wav_file) {
+    return {wav_file->sample_rate(), wav_file->num_channels()};
+  }
+  return {};
+}
+
 }  // namespace
 
 int main(int argc, char* argv[]) {
   google::SetUsageMessage(kUsage);
   google::ParseCommandLineFlags(&argc, &argv, true);
 
   if (!((FLAGS_i.empty()) ^ (FLAGS_dump.empty()))) {
     fprintf(stderr,
             "An input file must be specified with either -i or -dump.\n");
     return 1;
@@ skipping 79 matching lines @@
     out_rev_buf.reset(new ChannelBuffer<float>(
         rtc::CheckedDivExact(out_rev_file->sample_rate(), kChunksPerSecond),
         out_rev_file->num_channels()));
     out_rev_interleaved.resize(out_rev_buf->size());
   }
 
   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 auto input_config = MakeStreamConfig(&in_file);
+  const auto output_config = MakeStreamConfig(&out_file);
+  const auto reverse_input_config = MakeStreamConfig(in_rev_file.get());
+  const auto reverse_output_config = MakeStreamConfig(out_rev_file.get());
 
-  StreamConfig reverse_input_config = {};
-  StreamConfig reverse_output_config = {};
-  if (process_reverse) {
-    reverse_input_config = {in_rev_file->sample_rate(),
-                            in_rev_file->num_channels()};
-    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);
 }