VideoSendStream SSRC test.

Verifies that the VideoSendStream starts sending the set SSRC over RTP.

BUG=2227
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2074004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4573 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/video_engine/test/common/null_transport.cc

@@ -0,0 +1,24 @@
+/*
+ *  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 "webrtc/video_engine/test/common/null_transport.h"
+
+namespace webrtc {
+namespace test {
+
+bool NullTransport::SendRTP(const uint8_t* packet, size_t length) {
+  return true;
+}
+
+bool NullTransport::SendRTCP(const uint8_t* packet, size_t length) {
+  return true;
+}
+
+}  // namespace test
+}  // namespace webrtc

webrtc/video_engine/test/common/null_transport.h

@@ -0,0 +1,30 @@
+/*
+ *  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.
+ */
+#ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_NULL_TRANSPORT_H_
+#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_NULL_TRANSPORT_H_
+
+#include "webrtc/video_engine/new_include/transport.h"
+
+namespace webrtc {
+
+namespace newapi {
+class PacketReceiver;
+}  // namespace newapi
+
+namespace test {
+class NullTransport : public newapi::Transport {
+ public:
+  virtual bool SendRTP(const uint8_t* packet, size_t length) OVERRIDE;
+  virtual bool SendRTCP(const uint8_t* packet, size_t length) OVERRIDE;
+};
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // WEBRTC_VIDEO_ENGINE_TEST_COMMON_NULL_TRANSPORT_H_

webrtc/video_engine/test/send_stream_tests.cc

@@ -0,0 +1,81 @@
+/*
+ *  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 "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/system_wrappers/interface/event_wrapper.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/video_engine/test/common/frame_generator.h"
+#include "webrtc/video_engine/test/common/frame_generator_capturer.h"
+#include "webrtc/video_engine/test/common/null_transport.h"
+#include "webrtc/video_engine/new_include/video_call.h"
+#include "webrtc/video_engine/new_include/video_send_stream.h"
+
+namespace webrtc {
+
+class VideoSendStreamTest : public ::testing::Test {};
+class SendTransportObserver : public test::NullTransport {
+ public:
+  explicit SendTransportObserver(unsigned long timeout_ms)
+      : rtp_header_parser_(RtpHeaderParser::Create()),
+        send_test_complete_(EventWrapper::Create()),
+        timeout_ms_(timeout_ms) {}
+
+  EventTypeWrapper Wait() {
+    return send_test_complete_->Wait(timeout_ms_);
+  }
+
+ protected:
+  scoped_ptr<RtpHeaderParser> rtp_header_parser_;
+  scoped_ptr<EventWrapper> send_test_complete_;
+
+ private:
+  unsigned long timeout_ms_;
+};
+
+TEST_F(VideoSendStreamTest, SendsSetSsrc) {
+  static const uint32_t kSendSsrc = 0xC0FFEE;
+  class SendSsrcObserver : public SendTransportObserver {
+   public:
+    SendSsrcObserver() : SendTransportObserver(30 * 1000) {}
+
+    virtual bool SendRTP(const uint8_t* packet, size_t length) OVERRIDE {
+      RTPHeader header;
+      EXPECT_TRUE(
+          rtp_header_parser_->Parse(packet, static_cast<int>(length), &header));
+
+      if (header.ssrc == kSendSsrc)
+        send_test_complete_->Set();
+
+      return true;
+    }
+  } observer;
+
+  newapi::VideoCall::Config call_config(&observer);
+  scoped_ptr<newapi::VideoCall> call(newapi::VideoCall::Create(call_config));
+
+  newapi::VideoSendStream::Config send_config = call->GetDefaultSendConfig();
+  send_config.rtp.ssrcs.push_back(kSendSsrc);
+  newapi::VideoSendStream* send_stream = call->CreateSendStream(send_config);
+  scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer(
+      test::FrameGeneratorCapturer::Create(
+          send_stream->Input(),
+          test::FrameGenerator::Create(320, 240, Clock::GetRealTimeClock()),
+          30));
+  send_stream->StartSend();
+  frame_generator_capturer->Start();
+
+  EXPECT_EQ(kEventSignaled, observer.Wait());
+
+  frame_generator_capturer->Stop();
+  send_stream->StopSend();
+  call->DestroySendStream(send_stream);
+}
+
+}  // namespace webrtc

webrtc/video_engine/test/tests.gypi

@@ -32,6 +32,8 @@
         'common/mac/video_renderer_mac.h',
         'common/mac/video_renderer_mac.mm',
         'common/null_platform_renderer.cc',
+        'common/null_transport.cc',
+        'common/null_transport.h',
         'common/rtp_rtcp_observer.h',
         'common/run_tests.cc',
         'common/run_tests.h',
@@ -144,6 +146,7 @@
       'type': 'executable',
       'sources': [
         'engine_tests.cc',
+        'send_stream_tests.cc',
         'test_main.cc',
       ],
       'dependencies': [