Change from WebRtcRTPHeader to RTPHeader in NetEq tests and tools

webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.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.
  */
 
 #include "webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h"
 
 #include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/base/checks.h"
+#include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/typedefs.h"
 
 using webrtc::NetEq;
 using webrtc::test::AudioLoop;
 using webrtc::test::RtpGenerator;
-using webrtc::WebRtcRTPHeader;
 
 namespace webrtc {
 namespace test {
 
 int64_t NetEqPerformanceTest::Run(int runtime_ms,
                                   int lossrate,
                                   double drift_factor) {
   const std::string kInputFileName =
       webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
   const int kSampRateHz = 32000;
@@ skipping 14 matching lines @@
   AudioLoop audio_loop;
   const size_t kMaxLoopLengthSamples = kSampRateHz * 10;  // 10 second loop.
   const size_t kInputBlockSizeSamples = 60 * kSampRateHz / 1000;  // 60 ms.
   if (!audio_loop.Init(kInputFileName, kMaxLoopLengthSamples,
                        kInputBlockSizeSamples))
     return -1;
 
   int32_t time_now_ms = 0;
 
   // Get first input packet.
-  WebRtcRTPHeader rtp_header;
+  RTPHeader rtp_header;
   RtpGenerator rtp_gen(kSampRateHz / 1000);
   // Start with positive drift first half of simulation.
   rtp_gen.set_drift_factor(drift_factor);
   bool drift_flipped = false;
   int32_t packet_input_time_ms =
       rtp_gen.GetRtpHeader(kPayloadType, kInputBlockSizeSamples, &rtp_header);
   auto input_samples = audio_loop.GetNextBlock();
   if (input_samples.empty())
     exit(1);
   uint8_t input_payload[kInputBlockSizeSamples * sizeof(int16_t)];
   size_t payload_len = WebRtcPcm16b_Encode(input_samples.data(),
                                            input_samples.size(), input_payload);
   RTC_CHECK_EQ(sizeof(input_payload), payload_len);
 
   // Main loop.
   webrtc::Clock* clock = webrtc::Clock::GetRealTimeClock();
   int64_t start_time_ms = clock->TimeInMilliseconds();
   AudioFrame out_frame;
   while (time_now_ms < runtime_ms) {
     while (packet_input_time_ms <= time_now_ms) {
       // Drop every N packets, where N = FLAGS_lossrate.
       bool lost = false;
       if (lossrate > 0) {
-        lost = ((rtp_header.header.sequenceNumber - 1) % lossrate) == 0;
+        lost = ((rtp_header.sequenceNumber - 1) % lossrate) == 0;
       }
       if (!lost) {
         // Insert packet.
         int error =
-            neteq->InsertPacket(rtp_header.header, input_payload,
+            neteq->InsertPacket(rtp_header, input_payload,
                                 packet_input_time_ms * kSampRateHz / 1000);
         if (error != NetEq::kOK)
           return -1;
       }
 
       // Get next packet.
       packet_input_time_ms = rtp_gen.GetRtpHeader(kPayloadType,
                                                   kInputBlockSizeSamples,
                                                   &rtp_header);
       input_samples = audio_loop.GetNextBlock();
@@ skipping 22 matching lines @@
       drift_flipped = true;
     }
   }
   int64_t end_time_ms = clock->TimeInMilliseconds();
   delete neteq;
   return end_time_ms - start_time_ms;
 }
 
 }  // namespace test
 }  // namespace webrtc