2bb1bdab8d
A restarted VideoSendStream would previously be completely reset, causing gaps in sequence numbers and potentially RTP timestamps as well. This broke SRTP which requires fairly sequential sequence numbers. Presumably, were this sent without SRTP, we'd still have problems on the receiving end as the corresponding receiver is unaware of this reset. Also adding annotation to RTPSender and addressing some unlocked access to ssrc_, ssrc_rtx_ and rtx_. BUG= R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20819004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6612 4adac7df-926f-26a2-2b94-8c16560cd09d
124 lines
3.6 KiB
C++
124 lines
3.6 KiB
C++
/*
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* Copyright (c) 2014 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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#ifndef WEBRTC_TEST_COMMON_CALL_TEST_H_
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#define WEBRTC_TEST_COMMON_CALL_TEST_H_
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#include <vector>
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#include "webrtc/call.h"
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#include "webrtc/test/fake_decoder.h"
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#include "webrtc/test/fake_encoder.h"
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#include "webrtc/test/frame_generator_capturer.h"
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#include "webrtc/test/rtp_rtcp_observer.h"
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namespace webrtc {
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namespace test {
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class BaseTest;
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class CallTest : public ::testing::Test {
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public:
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CallTest();
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~CallTest();
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static const size_t kNumSsrcs = 3;
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static const unsigned int kDefaultTimeoutMs;
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static const unsigned int kLongTimeoutMs;
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static const uint8_t kSendPayloadType;
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static const uint8_t kSendRtxPayloadType;
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static const uint8_t kFakeSendPayloadType;
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static const uint32_t kSendRtxSsrcs[kNumSsrcs];
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static const uint32_t kSendSsrcs[kNumSsrcs];
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static const uint32_t kReceiverLocalSsrc;
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static const int kNackRtpHistoryMs;
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protected:
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void RunBaseTest(BaseTest* test);
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void CreateCalls(const Call::Config& sender_config,
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const Call::Config& receiver_config);
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void CreateSenderCall(const Call::Config& config);
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void CreateReceiverCall(const Call::Config& config);
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void CreateSendConfig(size_t num_streams);
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void CreateMatchingReceiveConfigs();
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void CreateFrameGeneratorCapturer();
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void CreateStreams();
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void Start();
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void Stop();
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void DestroyStreams();
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Clock* const clock_;
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scoped_ptr<Call> sender_call_;
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VideoSendStream::Config send_config_;
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std::vector<VideoStream> video_streams_;
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VideoSendStream* send_stream_;
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scoped_ptr<Call> receiver_call_;
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std::vector<VideoReceiveStream::Config> receive_configs_;
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std::vector<VideoReceiveStream*> receive_streams_;
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scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_;
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test::FakeEncoder fake_encoder_;
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test::FakeDecoder fake_decoder_;
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};
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class BaseTest : public RtpRtcpObserver {
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public:
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explicit BaseTest(unsigned int timeout_ms);
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BaseTest(unsigned int timeout_ms, const FakeNetworkPipe::Config& config);
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virtual ~BaseTest();
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virtual void PerformTest() = 0;
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virtual bool ShouldCreateReceivers() const = 0;
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virtual size_t GetNumStreams() const;
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virtual Call::Config GetSenderCallConfig();
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virtual Call::Config GetReceiverCallConfig();
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virtual void OnCallsCreated(Call* sender_call, Call* receiver_call);
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virtual void ModifyConfigs(
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VideoSendStream::Config* send_config,
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std::vector<VideoReceiveStream::Config>* receive_configs,
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std::vector<VideoStream>* video_streams);
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virtual void OnStreamsCreated(
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VideoSendStream* send_stream,
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const std::vector<VideoReceiveStream*>& receive_streams);
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virtual void OnFrameGeneratorCapturerCreated(
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FrameGeneratorCapturer* frame_generator_capturer);
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};
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class SendTest : public BaseTest {
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public:
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explicit SendTest(unsigned int timeout_ms);
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SendTest(unsigned int timeout_ms, const FakeNetworkPipe::Config& config);
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virtual bool ShouldCreateReceivers() const OVERRIDE;
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};
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class EndToEndTest : public BaseTest {
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public:
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explicit EndToEndTest(unsigned int timeout_ms);
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EndToEndTest(unsigned int timeout_ms, const FakeNetworkPipe::Config& config);
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virtual bool ShouldCreateReceivers() const OVERRIDE;
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};
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} // namespace test
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} // namespace webrtc
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#endif // WEBRTC_TEST_COMMON_CALL_TEST_H_
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