Add audio streams to CallTest and a first A/V call test.

webrtc/call/call_unittest.cc

 /*
  *  Copyright (c) 2015 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 <list>
 
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/audio_state.h"
 #include "webrtc/call.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/test/call_test.h"
 #include "webrtc/test/mock_voice_engine.h"
 
 namespace {
 
 struct CallHelper {
   CallHelper() {
     webrtc::AudioState::Config audio_state_config;
     audio_state_config.voice_engine = &voice_engine_;
     webrtc::Call::Config config;
     config.audio_state = webrtc::AudioState::Create(audio_state_config);
@@ skipping 72 matching lines @@
       } else {
         streams.push_front(stream);
       }
     }
     for (auto s : streams) {
       call->DestroyAudioReceiveStream(s);
     }
     streams.clear();
   }
 }
+
+namespace test {
+
+class MediumCallTest : public CallTest {};
+
+TEST_F(MediumCallTest, TransportSeqNumOnAudioAndVideo) {
+  class TransportSequenceNumberTest : public test::EndToEndTest {
+   public:
+    TransportSequenceNumberTest()
+        : EndToEndTest(test::CallTest::kDefaultTimeoutMs),
+          video_observed_(false),
+          audio_observed_(false) {
+      parser_->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber,
+                                          kExtensionId);
+    }
+
+    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) {
+      send_config->rtp.extensions.clear();
+      send_config->rtp.extensions.push_back(
+          RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));
+      (*receive_configs)[0].rtp.extensions.clear();
+      (*receive_configs)[0].rtp.extensions.push_back(
+          RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));
+    }
+
+    void ModifyAudioConfigs(
+        AudioSendStream::Config* send_config,
+        std::vector<AudioReceiveStream::Config>* receive_configs) {
+      send_config->rtp.extensions.clear();
+      send_config->rtp.extensions.push_back(
+          RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));
+      (*receive_configs)[0].rtp.extensions.clear();
+      (*receive_configs)[0].rtp.extensions.push_back(
+          RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));
+    }
+
+    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 =
+          unwrapper_.Unwrap(header.extension.transportSequenceNumber);
+      EXPECT_TRUE(received_packet_ids_.insert(packet_id).second);
+
+      if (header.ssrc == kVideoSendSsrcs[0])
+        video_observed_ = true;
+      if (header.ssrc == kAudioSendSsrc)
+        audio_observed_ = true;
+      if (audio_observed_ && video_observed_ &&
+          received_packet_ids_.size() == 50) {
+        size_t packet_id_range =
+            *received_packet_ids_.rbegin() - *received_packet_ids_.begin() + 1;
+        EXPECT_EQ(received_packet_ids_.size(), packet_id_range);
+        observation_complete_.Set();
+      }
+      return SEND_PACKET;
+    }
+
+    void PerformTest() override {
+      EXPECT_TRUE(Wait()) << "Timed out while waiting for audio and video "
+                             "packets with transport sequence number.";
+    }
+
+   private:
+    const int kExtensionId = 8;
+    bool video_observed_;
+    bool audio_observed_;
+    SequenceNumberUnwrapper unwrapper_;
+    std::set<int64_t> received_packet_ids_;
+  } test;
+
+  RunBaseTest(&test, FakeNetworkPipe::Config());

[pbos-webrtc, 2015/12/28 22:22:31]

Wondering if this could be confused being put under namespace test, maybe worth
figuring out?

[stefan-webrtc, 2015/12/29 09:37:06]

I moved it in the latest patch set. Still under test, but so are all of the
other tests in end_to_end_tests.cc. Oh, you are saying that "test" here may be
confused with "namespace test". Maybe :)

+}
+}  // namespace test
 }  // namespace webrtc