be9d2a4549
Allows setting SSRCs for future simulcast layers even though no set send codec uses them. Also re-enabling CanSwitchToUseAllSsrcs as an end-to-end test, required for bitrate ramp-up, instead of send-side only (resolving issue 3078). This test was used to verify reserved modules' SSRCs are preserved correctly. To enable a multiple-stream end-to-end test test::CallTest was modified to work on a vector of receive streams instead of just one. BUG=3078 R=kjellander@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/15859005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6565 4adac7df-926f-26a2-2b94-8c16560cd09d
122 lines
3.6 KiB
C++
122 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 kSendRtxSsrc;
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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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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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