Replace RTCPUtility RTCPParser with TestRtcpParser

webrtc/video/video_send_stream_tests.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.
  */
 #include <algorithm>  // max
 #include <memory>
 #include <vector>
 
 #include "webrtc/base/bind.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/event.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/platform_thread.h"
 #include "webrtc/base/rate_limiter.h"
 #include "webrtc/call.h"
 #include "webrtc/call/transport_adapter.h"
 #include "webrtc/common_video/include/frame_callback.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
-#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h"
 #include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h"
 #include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/test/call_test.h"
 #include "webrtc/test/configurable_frame_size_encoder.h"
 #include "webrtc/test/fake_texture_frame.h"
 #include "webrtc/test/frame_generator.h"
 #include "webrtc/test/frame_utils.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/null_transport.h"
+#include "webrtc/test/rtcp_packet_parser.h"
 #include "webrtc/test/testsupport/perf_test.h"
 
 #include "webrtc/video/send_statistics_proxy.h"
 #include "webrtc/video_frame.h"
 #include "webrtc/video_send_stream.h"
 
 namespace webrtc {
 
 enum VideoFormat { kGeneric, kVP8, };
 
@@ skipping 34 matching lines @@
 }
 
 TEST_F(VideoSendStreamTest, SupportsCName) {
   static std::string kCName = "PjQatC14dGfbVwGPUOA9IH7RlsFDbWl4AhXEiDsBizo=";
   class CNameObserver : public test::SendTest {
    public:
     CNameObserver() : SendTest(kDefaultTimeoutMs) {}
 
    private:
     Action OnSendRtcp(const uint8_t* packet, size_t length) override {
-      RTCPUtility::RTCPParserV2 parser(packet, length, true);
-      EXPECT_TRUE(parser.IsValid());
+      test::RtcpPacketParser parser;
+      EXPECT_TRUE(parser.Parse(packet, length));
+      if (parser.sdes()->num_packets() > 0) {
+        EXPECT_EQ(1u, parser.sdes()->chunks().size());
+        EXPECT_EQ(kCName, parser.sdes()->chunks()[0].cname);
 
-      RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
-      while (packet_type != RTCPUtility::RTCPPacketTypes::kInvalid) {
-        if (packet_type == RTCPUtility::RTCPPacketTypes::kSdesChunk) {
-          EXPECT_EQ(parser.Packet().CName.CName, kCName);
-          observation_complete_.Set();
-        }
-
-        packet_type = parser.Iterate();
+        observation_complete_.Set();
       }
 
       return SEND_PACKET;
     }
 
     void ModifyVideoConfigs(
         VideoSendStream::Config* send_config,
         std::vector<VideoReceiveStream::Config>* receive_configs,
         VideoEncoderConfig* encoder_config) override {
       send_config->rtp.c_name = kCName;
@@ skipping 1962 matching lines @@
       rtc::CritScope lock(&crit_);
       RTPHeader header;
       EXPECT_TRUE(parser_->Parse(packet, length, &header));
       ++rtp_packets_sent_;
       media_bytes_sent_ += length - header.headerLength - header.paddingLength;
       return SEND_PACKET;
     }
 
     Action OnSendRtcp(const uint8_t* packet, size_t length) override {
       rtc::CritScope lock(&crit_);
-      RTCPUtility::RTCPParserV2 parser(packet, length, true);
-      EXPECT_TRUE(parser.IsValid());
+      test::RtcpPacketParser parser;
+      EXPECT_TRUE(parser.Parse(packet, length));
 
-      RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
-      while (packet_type != RTCPUtility::RTCPPacketTypes::kInvalid) {
-        if (packet_type == RTCPUtility::RTCPPacketTypes::kSr) {
-          // Only compare sent media bytes if SenderPacketCount matches the
-          // number of sent rtp packets (a new rtp packet could be sent before
-          // the rtcp packet).
-          if (parser.Packet().SR.SenderOctetCount > 0 &&
-              parser.Packet().SR.SenderPacketCount == rtp_packets_sent_) {
-            EXPECT_EQ(media_bytes_sent_, parser.Packet().SR.SenderOctetCount);
-            observation_complete_.Set();
-          }
+      if (parser.sender_report()->num_packets() > 0) {
+        // Only compare sent media bytes if SenderPacketCount matches the
+        // number of sent rtp packets (a new rtp packet could be sent before
+        // the rtcp packet).
+        if (parser.sender_report()->sender_octet_count() > 0 &&
+            parser.sender_report()->sender_packet_count() ==
+                rtp_packets_sent_) {
+          EXPECT_EQ(media_bytes_sent_,
+                    parser.sender_report()->sender_octet_count());
+          observation_complete_.Set();
         }
-        packet_type = parser.Iterate();
       }
 
       return SEND_PACKET;
     }
 
     void PerformTest() override {
       EXPECT_TRUE(Wait()) << "Timed out while waiting for RTCP sender report.";
     }
 
     rtc::CriticalSection crit_;
@@ skipping 762 matching lines @@
         observation_complete_.Set();
       }
     }
   } test;
 
   RunBaseTest(&test);
 }
 #endif  // !defined(RTC_DISABLE_VP9)
 
 }  // namespace webrtc