Revert of Allow more than 2 input channels in AudioProcessing.

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 109 matching lines @@
       in_file.num_channels());
   ChannelBuffer<float> out_buf(
       rtc::CheckedDivExact(out_file.sample_rate(), kChunksPerSecond),
       out_file.num_channels());
 
   std::vector<float> in_interleaved(in_buf.size());
   std::vector<float> out_interleaved(out_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(),
-  };
   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 (FLAGS_perf) {
       processing_start_time = TickTime::Now();
     }
-    CHECK_EQ(kNoErr, ap->ProcessStream(in_buf.channels(), input_config,
-                                       output_config, out_buf.channels()));
+    CHECK_EQ(kNoErr,
+        ap->ProcessStream(in_buf.channels(),
+                          in_buf.num_frames(),
+                          in_file.sample_rate(),
+                          LayoutFromChannels(in_buf.num_channels()),
+                          out_file.sample_rate(),
+                          LayoutFromChannels(out_buf.num_channels()),
+                          out_buf.channels()));
     if (FLAGS_perf) {
       accumulated_time += TickTime::Now() - processing_start_time;
     }
 
     Interleave(out_buf.channels(), out_buf.num_frames(),
                out_buf.num_channels(), &out_interleaved[0]);
     FloatToFloatS16(&out_interleaved[0], out_interleaved.size(),
                     &out_interleaved[0]);
     out_file.WriteSamples(&out_interleaved[0], out_interleaved.size());
     num_chunks++;
   }
   if (FLAGS_perf) {
     int64_t execution_time_ms = accumulated_time.Milliseconds();
     printf("\nExecution time: %.3f s\nFile time: %.2f s\n"
            "Time per chunk: %.3f ms\n",
            execution_time_ms * 0.001f, num_chunks * 1.f / kChunksPerSecond,
            execution_time_ms * 1.f / num_chunks);
   }
   return 0;
 }
 
 }  // namespace webrtc
 
 int main(int argc, char* argv[]) {
   return webrtc::main(argc, argv);
 }