Change NetEq::GetPlayoutTimestamp to return an rtc::Optional<uint32_t>

webrtc/modules/audio_coding/test/delay_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 162 matching lines @@
     out_file_b_.Open(file_name.c_str(), 32000, "wb");
   }
 
   void Run(int duration_sec, const char* output_prefix) {
     OpenOutFile(output_prefix);
     AudioFrame audio_frame;
     uint32_t out_freq_hz_b = out_file_b_.SamplingFrequency();
 
     int num_frames = 0;
     int in_file_frames = 0;
-    uint32_t playout_ts;
     uint32_t received_ts;
     double average_delay = 0;
     double inst_delay_sec = 0;
     while (num_frames < (duration_sec * 100)) {
       if (in_file_a_.EndOfFile()) {
         in_file_a_.Rewind();
       }
 
       // Print delay information every 16 frame
       if ((num_frames & 0x3F) == 0x3F) {
         NetworkStatistics statistics;
         acm_b_->GetNetworkStatistics(&statistics);
         fprintf(stdout, "delay: min=%3d  max=%3d  mean=%3d  median=%3d"
                 " ts-based average = %6.3f, "
                 "curr buff-lev = %4u opt buff-lev = %4u \n",
                 statistics.minWaitingTimeMs, statistics.maxWaitingTimeMs,
                 statistics.meanWaitingTimeMs, statistics.medianWaitingTimeMs,
                 average_delay, statistics.currentBufferSize,
                 statistics.preferredBufferSize);
         fflush (stdout);
       }
 
       in_file_a_.Read10MsData(audio_frame);
       ASSERT_GE(acm_a_->Add10MsData(audio_frame), 0);
       ASSERT_EQ(0, acm_b_->PlayoutData10Ms(out_freq_hz_b, &audio_frame));
       out_file_b_.Write10MsData(
           audio_frame.data_,
           audio_frame.samples_per_channel_ * audio_frame.num_channels_);
-      acm_b_->PlayoutTimestamp(&playout_ts);
       received_ts = channel_a2b_->LastInTimestamp();
-      inst_delay_sec = static_cast<uint32_t>(received_ts - playout_ts)
-          / static_cast<double>(encoding_sample_rate_hz_);
+      rtc::Optional<uint32_t> playout_timestamp = acm_b_->PlayoutTimestamp();
+      ASSERT_TRUE(playout_timestamp);
+      inst_delay_sec = static_cast<uint32_t>(received_ts - *playout_timestamp) /
+                       static_cast<double>(encoding_sample_rate_hz_);
 
       if (num_frames > 10)
         average_delay = 0.95 * average_delay + 0.05 * inst_delay_sec;
 
       ++num_frames;
       ++in_file_frames;
     }
     out_file_b_.Close();
   }
 
@@ skipping 30 matching lines @@
   test_setting.codec.num_channels = FLAGS_num_channels;
   test_setting.acm.dtx = FLAGS_dtx;
   test_setting.acm.fec = FLAGS_fec;
   test_setting.packet_loss = FLAGS_packet_loss;
 
   webrtc::DelayTest delay_test;
   delay_test.Initialize();
   delay_test.Perform(&test_setting, 1, 240, "delay_test");
   return 0;
 }