Remove use of RtpHeaderExtension and clean up

webrtc/video/end_to_end_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>
@@ skipping 1414 matching lines @@
       RTC_DCHECK(observer_);
       EXPECT_TRUE(observer_->Wait());
     }
 
     void UpdateSendConfig(
         size_t stream_index,
         VideoSendStream::Config* send_config,
         VideoEncoderConfig* encoder_config,
         test::FrameGeneratorCapturer** frame_generator) override {
       send_config->rtp.extensions.clear();
-      send_config->rtp.extensions.push_back(
-          RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));
+      send_config->rtp.extensions.push_back(RtpExtension(
+          RtpExtension::kTransportSequenceNumberUri, kExtensionId));
 
       // Force some padding to be sent.
       const int kPaddingBitrateBps = 50000;
       int total_target_bitrate = 0;
       for (const VideoStream& stream : encoder_config->streams)
         total_target_bitrate += stream.target_bitrate_bps;
       encoder_config->min_transmit_bitrate_bps =
           total_target_bitrate + kPaddingBitrateBps;
 
       // Configure RTX for redundant payload padding.
       send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
       send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[stream_index]);
       send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
       rtx_to_media_ssrcs_[kSendRtxSsrcs[stream_index]] =
           send_config->rtp.ssrcs[0];
 
       if (stream_index == 0)
         first_media_ssrc_ = send_config->rtp.ssrcs[0];
     }
 
     void UpdateReceiveConfig(
         size_t stream_index,
         VideoReceiveStream::Config* receive_config) override {
       receive_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
       receive_config->rtp.extensions.clear();
-      receive_config->rtp.extensions.push_back(
-          RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));
+      receive_config->rtp.extensions.push_back(RtpExtension(
+          RtpExtension::kTransportSequenceNumberUri, kExtensionId));
     }
 
     test::DirectTransport* CreateSendTransport(Call* sender_call) override {
       observer_ = new RtpExtensionHeaderObserver(sender_call, first_media_ssrc_,
                                                  rtx_to_media_ssrcs_);
       return observer_;
     }
 
    private:
     uint32_t first_media_ssrc_;
@@ skipping 58 matching lines @@
 
   size_t GetNumVideoStreams() const override { return num_video_streams_; }
   size_t GetNumAudioStreams() const override { return num_audio_streams_; }
 
   void ModifyVideoConfigs(
       VideoSendStream::Config* send_config,
       std::vector<VideoReceiveStream::Config>* receive_configs,
       VideoEncoderConfig* encoder_config) override {
     send_config->rtp.extensions.clear();
     send_config->rtp.extensions.push_back(
-        RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));
+        RtpExtension(RtpExtension::kTransportSequenceNumberUri, kExtensionId));
     (*receive_configs)[0].rtp.extensions = send_config->rtp.extensions;
     (*receive_configs)[0].rtp.transport_cc = feedback_enabled_;
   }
 
   void ModifyAudioConfigs(
       AudioSendStream::Config* send_config,
       std::vector<AudioReceiveStream::Config>* receive_configs) override {
     send_config->rtp.extensions.clear();
     send_config->rtp.extensions.push_back(
-        RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));
+        RtpExtension(RtpExtension::kTransportSequenceNumberUri, kExtensionId));
     (*receive_configs)[0].rtp.extensions.clear();
     (*receive_configs)[0].rtp.extensions = send_config->rtp.extensions;
     (*receive_configs)[0].rtp.transport_cc = feedback_enabled_;
   }
 
  private:
   static const int kExtensionId = 5;
   const bool feedback_enabled_;
   const size_t num_video_streams_;
   const size_t num_audio_streams_;
@@ skipping 217 matching lines @@
       return config;
     }
 
     void ModifyVideoConfigs(
         VideoSendStream::Config* send_config,
         std::vector<VideoReceiveStream::Config>* receive_configs,
         VideoEncoderConfig* encoder_config) override {
       ASSERT_EQ(1u, send_config->rtp.ssrcs.size());
       send_config->rtp.extensions.clear();
       send_config->rtp.extensions.push_back(
-          RtpExtension(RtpExtension::kTransportSequenceNumber,
+          RtpExtension(RtpExtension::kTransportSequenceNumberUri,
                        test::kTransportSequenceNumberExtensionId));
       sender_ssrc_ = send_config->rtp.ssrcs[0];
 
       encoder_config->streams[0].max_bitrate_bps =
           encoder_config->streams[0].target_bitrate_bps = 2000000;
 
       ASSERT_EQ(1u, receive_configs->size());
       (*receive_configs)[0].rtp.remb = false;
       (*receive_configs)[0].rtp.transport_cc = true;
       (*receive_configs)[0].rtp.extensions = send_config->rtp.extensions;
@@ skipping 1644 matching lines @@
     }
 
     size_t GetNumVideoStreams() const override { return 1; }
     size_t GetNumAudioStreams() const override { return 1; }
 
     void ModifyVideoConfigs(
         VideoSendStream::Config* send_config,
         std::vector<VideoReceiveStream::Config>* receive_configs,
         VideoEncoderConfig* encoder_config) override {
       send_config->rtp.extensions.clear();
-      send_config->rtp.extensions.push_back(
-          RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));
+      send_config->rtp.extensions.push_back(RtpExtension(
+          RtpExtension::kTransportSequenceNumberUri, kExtensionId));
       (*receive_configs)[0].rtp.extensions = send_config->rtp.extensions;
     }
 
     void ModifyAudioConfigs(
         AudioSendStream::Config* send_config,
         std::vector<AudioReceiveStream::Config>* receive_configs) override {
       send_config->rtp.extensions.clear();
-      send_config->rtp.extensions.push_back(
-          RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));
+      send_config->rtp.extensions.push_back(RtpExtension(
+          RtpExtension::kTransportSequenceNumberUri, kExtensionId));
       (*receive_configs)[0].rtp.extensions.clear();
       (*receive_configs)[0].rtp.extensions = send_config->rtp.extensions;
     }
 
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
       RTPHeader header;
       EXPECT_TRUE(parser_->Parse(packet, length, &header));
       EXPECT_TRUE(header.extension.hasTransportSequenceNumber);
       // Unwrap packet id and verify uniqueness.
       int64_t packet_id =
@@ skipping 22 matching lines @@
    private:
     bool video_observed_;
     bool audio_observed_;
     SequenceNumberUnwrapper unwrapper_;
     std::set<int64_t> received_packet_ids_;
   } test;
 
   RunBaseTest(&test);
 }
 }  // namespace webrtc