Reduce the runtime of some ACM tests in modules_tests

webrtc/modules/audio_coding/test/TestStereo.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.
  */
 
 #include "webrtc/modules/audio_coding/test/TestStereo.h"
 
 #include <assert.h>
 
 #include <string>
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/checks.h"
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
 #include "webrtc/modules/audio_coding/test/utility.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
 
 // Class for simulating packet handling
@@ skipping 700 matching lines @@
                      int percent_loss) {
   AudioFrame audio_frame;
 
   int32_t out_freq_hz_b = out_file_.SamplingFrequency();
   uint16_t rec_size;
   uint32_t time_stamp_diff;
   channel->reset_payload_size();
   int error_count = 0;
   int variable_bytes = 0;
   int variable_packets = 0;
+  static const int kTestLengthMs = 500;
+  const int test_length_blocks = rtc::CheckedDivExact(kTestLengthMs, 10);
+  in_file_mono_->SetNum10MsBlocksToRead(test_length_blocks);
+  in_file_stereo_->SetNum10MsBlocksToRead(test_length_blocks);
+  // Fast-forward 1 second (100 blocks) since the files start with silence.
+  in_file_stereo_->FastForward(100);
+  in_file_mono_->FastForward(100);
 
   while (1) {
     // Simulate packet loss by setting |packet_loss_| to "true" in
     // |percent_loss| percent of the loops.
     if (percent_loss > 0) {
       if (counter_ == floor((100 / percent_loss) + 0.5)) {
         counter_ = 0;
         channel->set_lost_packet(true);
       } else {
         channel->set_lost_packet(false);
@@ skipping 45 matching lines @@
         audio_frame.data_,
         audio_frame.samples_per_channel_ * audio_frame.num_channels_);
   }
 
   EXPECT_EQ(0, error_count);
 
   // Check that packet size is in the right range for variable rate codecs,
   // such as Opus.
   if (variable_packets > 0) {
     variable_bytes /= variable_packets;
-    EXPECT_NEAR(variable_bytes, pack_size_bytes_, 3);
+    EXPECT_NEAR(variable_bytes, pack_size_bytes_, 18);
   }
 
   if (in_file_mono_->EndOfFile()) {
     in_file_mono_->Rewind();
   }
   if (in_file_stereo_->EndOfFile()) {
     in_file_stereo_->Rewind();
   }
   // Reset in case we ended with a lost packet
   channel->set_lost_packet(false);
@@ skipping 15 matching lines @@
     printf("%s -> ", send_codec->plname);
   }
   CodecInst receive_codec;
   acm_b_->ReceiveCodec(&receive_codec);
   if (test_mode_ != 0) {
     printf("%s\n", receive_codec.plname);
   }
 }
 
 }  // namespace webrtc