Reduce the runtime of some ACM tests in modules_tests

webrtc/modules/audio_coding/test/opus_test.cc

 /*
  *  Copyright (c) 2013 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 217 matching lines @@
 
 #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_ARCH_ARM)
   // If we are on Android, iOS and/or ARM, use a lower complexity setting as
   // default.
   const int kOpusComplexity5 = 5;
   EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_mono_encoder_, kOpusComplexity5));
   EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_stereo_encoder_,
                                         kOpusComplexity5));
 #endif
 
-  // Make sure the runtime is less than 60 seconds to pass Android test.
-  for (size_t audio_length = 0; audio_length < 10000; audio_length += 10) {
+  // Fast-forward 1 second (100 blocks) since the files start with silence.
+  in_file_stereo_.FastForward(100);
+  in_file_mono_.FastForward(100);
+
+  // Limit the runtime to 1000 blocks of 10 ms each.
+  for (size_t audio_length = 0; audio_length < 1000; audio_length += 10) {
     bool lost_packet = false;
 
     // Get 10 msec of audio.
     if (channels == 1) {
       if (in_file_mono_.EndOfFile()) {
         break;
       }
       in_file_mono_.Read10MsData(audio_frame);
     } else {
       if (in_file_stereo_.EndOfFile()) {
@@ skipping 121 matching lines @@
   out_file_.Open(file_name, 48000, "wb");
   file_stream.str("");
   file_name = file_stream.str();
   file_stream << webrtc::test::OutputPath() << "opusstandalone_out_"
       << test_number << ".pcm";
   file_name = file_stream.str();
   out_file_standalone_.Open(file_name, 48000, "wb");
 }
 
 }  // namespace webrtc