Implement dual stream full stack test and loopback tool
Bug: webrtc:8588 Change-Id: I0abec4891a723c98001f4580f0cfa57a5d6d6bdb Reviewed-on: https://webrtc-review.googlesource.com/34441 Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21416}
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@ -38,7 +38,8 @@ class CallTest : public ::testing::Test {
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CallTest();
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virtual ~CallTest();
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static const size_t kNumSsrcs = 3;
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static constexpr size_t kNumSsrcs = 6;
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static const int kNumSimulcastStreams = 3;
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static const int kDefaultWidth = 320;
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static const int kDefaultHeight = 180;
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static const int kDefaultFramerate = 30;
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@ -77,11 +78,22 @@ class CallTest : public ::testing::Test {
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void CreateReceiverCall(const Call::Config& config);
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void DestroyCalls();
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void CreateVideoSendConfig(VideoSendStream::Config* video_config,
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size_t num_video_streams,
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size_t num_used_ssrcs,
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Transport* send_transport);
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void CreateAudioAndFecSendConfigs(size_t num_audio_streams,
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size_t num_flexfec_streams,
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Transport* send_transport);
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void CreateSendConfig(size_t num_video_streams,
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size_t num_audio_streams,
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size_t num_flexfec_streams,
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Transport* send_transport);
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std::vector<VideoReceiveStream::Config> CreateMatchingVideoReceiveConfigs(
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const VideoSendStream::Config& video_send_config,
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Transport* rtcp_send_transport);
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void CreateMatchingAudioAndFecConfigs(Transport* rtcp_send_transport);
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void CreateMatchingReceiveConfigs(Transport* rtcp_send_transport);
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void CreateFrameGeneratorCapturerWithDrift(Clock* drift_clock,
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