Reduce the runtime of some ACM tests in modules_tests

webrtc/modules/audio_coding/test/TestAllCodecs.cc

 /*
  *  Copyright (c) 2012 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 404 matching lines @@
 }
 
 void TestAllCodecs::Run(TestPack* channel) {
   AudioFrame audio_frame;
 
   int32_t out_freq_hz = outfile_b_.SamplingFrequency();
   size_t receive_size;
   uint32_t timestamp_diff;
   channel->reset_payload_size();
   int error_count = 0;
+  int counter = 0;
+  // Set test length to 500 ms (50 blocks of 10 ms each).
+  infile_a_.SetNum10MsBlocksToRead(50);
+  // Fast-forward 1 second (100 blocks) since the file starts with silence.
+  infile_a_.FastForward(100);
 
-  int counter = 0;
   while (!infile_a_.EndOfFile()) {
     // Add 10 msec to ACM.
     infile_a_.Read10MsData(audio_frame);
     CHECK_ERROR(acm_a_->Add10MsData(audio_frame));
 
     // Verify that the received packet size matches the settings.
     receive_size = channel->payload_size();
     if (receive_size) {
       if ((receive_size != packet_size_bytes_) &&
           (packet_size_bytes_ != kVariableSize)) {
@@ skipping 39 matching lines @@
 }
 
 void TestAllCodecs::DisplaySendReceiveCodec() {
   CodecInst my_codec_param;
   printf("%s -> ", acm_a_->SendCodec()->plname);
   acm_b_->ReceiveCodec(&my_codec_param);
   printf("%s\n", my_codec_param.plname);
 }
 
 }  // namespace webrtc