1 files changed, 92 insertions, 0 deletions
diff --git a/third_party/libwebrtc/video/end_to_end_tests/multi_stream_tests.cc b/third_party/libwebrtc/video/end_to_end_tests/multi_stream_tests.cc
new file mode 100644
index 0000000000..b997538d96
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+++ b/third_party/libwebrtc/video/end_to_end_tests/multi_stream_tests.cc
@@ -0,0 +1,92 @@
+/*
+ *  Copyright 2018 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 <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+#include <vector>
+
+#include "api/video/video_frame.h"
+#include "api/video/video_sink_interface.h"
+#include "call/rtp_config.h"
+#include "call/video_receive_stream.h"
+#include "call/video_send_stream.h"
+#include "rtc_base/event.h"
+#include "test/frame_generator_capturer.h"
+#include "test/gtest.h"
+#include "video/config/video_encoder_config.h"
+#include "video/end_to_end_tests/multi_stream_tester.h"
+
+namespace webrtc {
+// Each renderer verifies that it receives the expected resolution, and as soon
+// as every renderer has received a frame, the test finishes.
+TEST(MultiStreamEndToEndTest, SendsAndReceivesMultipleStreams) {
+  class VideoOutputObserver : public rtc::VideoSinkInterface<VideoFrame> {
+   public:
+    VideoOutputObserver(const MultiStreamTester::CodecSettings& settings,
+                        uint32_t ssrc,
+                        test::FrameGeneratorCapturer** frame_generator)
+        : settings_(settings), ssrc_(ssrc), frame_generator_(frame_generator) {}
+
+    void OnFrame(const VideoFrame& video_frame) override {
+      EXPECT_EQ(settings_.width, video_frame.width());
+      EXPECT_EQ(settings_.height, video_frame.height());
+      (*frame_generator_)->Stop();
+      done_.Set();
+    }
+
+    uint32_t Ssrc() { return ssrc_; }
+
+    bool Wait() { return done_.Wait(TimeDelta::Seconds(30)); }
+
+   private:
+    const MultiStreamTester::CodecSettings& settings_;
+    const uint32_t ssrc_;
+    test::FrameGeneratorCapturer** const frame_generator_;
+    rtc::Event done_;
+  };
+
+  class Tester : public MultiStreamTester {
+   public:
+    Tester() = default;
+    ~Tester() override = default;
+
+   protected:
+    void Wait() override {
+      for (const auto& observer : observers_) {
+        EXPECT_TRUE(observer->Wait())
+            << "Time out waiting for from on ssrc " << observer->Ssrc();
+      }
+    }
+
+    void UpdateSendConfig(
+        size_t stream_index,
+        VideoSendStream::Config* send_config,
+        VideoEncoderConfig* encoder_config,
+        test::FrameGeneratorCapturer** frame_generator) override {
+      observers_[stream_index] = std::make_unique<VideoOutputObserver>(
+          codec_settings[stream_index], send_config->rtp.ssrcs.front(),
+          frame_generator);
+    }
+
+    void UpdateReceiveConfig(
+        size_t stream_index,
+        VideoReceiveStreamInterface::Config* receive_config) override {
+      receive_config->renderer = observers_[stream_index].get();
+    }
+
+   private:
+    std::unique_ptr<VideoOutputObserver> observers_[kNumStreams];
+  } tester;
+
+  tester.RunTest();
+}
+}  // namespace webrtc