74fa4893f9
Typing newapi:: everywhere is very verbose, and doesn't add any real value. The new API is still separated from other code by being in separate directories, such as internal/ or new_include. BUG= R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2075004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4601 4adac7df-926f-26a2-2b94-8c16560cd09d
128 lines
4.2 KiB
C++
128 lines
4.2 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
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#include "webrtc/system_wrappers/interface/event_wrapper.h"
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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#include "webrtc/video_engine/test/common/frame_generator.h"
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#include "webrtc/video_engine/test/common/frame_generator_capturer.h"
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#include "webrtc/video_engine/test/common/null_transport.h"
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#include "webrtc/video_engine/new_include/video_call.h"
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#include "webrtc/video_engine/new_include/video_send_stream.h"
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namespace webrtc {
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class SendTransportObserver : public test::NullTransport {
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public:
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explicit SendTransportObserver(unsigned long timeout_ms)
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: rtp_header_parser_(RtpHeaderParser::Create()),
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send_test_complete_(EventWrapper::Create()),
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timeout_ms_(timeout_ms) {}
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EventTypeWrapper Wait() {
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return send_test_complete_->Wait(timeout_ms_);
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}
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protected:
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scoped_ptr<RtpHeaderParser> rtp_header_parser_;
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scoped_ptr<EventWrapper> send_test_complete_;
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private:
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unsigned long timeout_ms_;
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};
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class VideoSendStreamTest : public ::testing::Test {
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protected:
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static const uint32_t kSendSsrc;
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void RunSendTest(VideoCall* call,
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const VideoSendStream::Config& config,
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SendTransportObserver* observer) {
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VideoSendStream* send_stream = call->CreateSendStream(config);
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scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer(
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test::FrameGeneratorCapturer::Create(
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send_stream->Input(),
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test::FrameGenerator::Create(320, 240, Clock::GetRealTimeClock()),
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30));
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send_stream->StartSend();
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frame_generator_capturer->Start();
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EXPECT_EQ(kEventSignaled, observer->Wait());
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frame_generator_capturer->Stop();
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send_stream->StopSend();
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call->DestroySendStream(send_stream);
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}
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};
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const uint32_t VideoSendStreamTest::kSendSsrc = 0xC0FFEE;
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TEST_F(VideoSendStreamTest, SendsSetSsrc) {
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class SendSsrcObserver : public SendTransportObserver {
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public:
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SendSsrcObserver() : SendTransportObserver(30 * 1000) {}
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virtual bool SendRTP(const uint8_t* packet, size_t length) OVERRIDE {
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RTPHeader header;
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EXPECT_TRUE(
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rtp_header_parser_->Parse(packet, static_cast<int>(length), &header));
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if (header.ssrc == kSendSsrc)
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send_test_complete_->Set();
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return true;
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}
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} observer;
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VideoCall::Config call_config(&observer);
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scoped_ptr<VideoCall> call(VideoCall::Create(call_config));
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VideoSendStream::Config send_config = call->GetDefaultSendConfig();
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send_config.rtp.ssrcs.push_back(kSendSsrc);
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RunSendTest(call.get(), send_config, &observer);
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}
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TEST_F(VideoSendStreamTest, SupportsCName) {
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static std::string kCName = "PjQatC14dGfbVwGPUOA9IH7RlsFDbWl4AhXEiDsBizo=";
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class CNameObserver : public SendTransportObserver {
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public:
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CNameObserver() : SendTransportObserver(30 * 1000) {}
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virtual bool SendRTCP(const uint8_t* packet, size_t length) OVERRIDE {
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RTCPUtility::RTCPParserV2 parser(packet, length, true);
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EXPECT_TRUE(parser.IsValid());
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RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
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while (packet_type != RTCPUtility::kRtcpNotValidCode) {
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if (packet_type == RTCPUtility::kRtcpSdesChunkCode) {
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EXPECT_EQ(parser.Packet().CName.CName, kCName);
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send_test_complete_->Set();
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}
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packet_type = parser.Iterate();
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}
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return true;
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}
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} observer;
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VideoCall::Config call_config(&observer);
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scoped_ptr<VideoCall> call(VideoCall::Create(call_config));
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VideoSendStream::Config send_config = call->GetDefaultSendConfig();
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send_config.rtp.ssrcs.push_back(kSendSsrc);
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send_config.rtp.c_name = kCName;
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RunSendTest(call.get(), send_config, &observer);
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}
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} // namespace webrtc
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