1 files changed, 79 insertions, 0 deletions
diff --git a/third_party/libwebrtc/test/layer_filtering_transport.h b/third_party/libwebrtc/test/layer_filtering_transport.h
new file mode 100644
index 0000000000..f4aa550d74
--- /dev/null
+++ b/third_party/libwebrtc/test/layer_filtering_transport.h
@@ -0,0 +1,79 @@
+/*
+ *  Copyright (c) 2015 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.
+ */
+#ifndef TEST_LAYER_FILTERING_TRANSPORT_H_
+#define TEST_LAYER_FILTERING_TRANSPORT_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <map>
+#include <memory>
+
+#include "api/call/transport.h"
+#include "api/media_types.h"
+#include "call/call.h"
+#include "call/simulated_packet_receiver.h"
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
+#include "test/direct_transport.h"
+
+namespace webrtc {
+
+namespace test {
+
+class LayerFilteringTransport : public test::DirectTransport {
+ public:
+  LayerFilteringTransport(
+      TaskQueueBase* task_queue,
+      std::unique_ptr<SimulatedPacketReceiverInterface> pipe,
+      Call* send_call,
+      uint8_t vp8_video_payload_type,
+      uint8_t vp9_video_payload_type,
+      int selected_tl,
+      int selected_sl,
+      const std::map<uint8_t, MediaType>& payload_type_map,
+      uint32_t ssrc_to_filter_min,
+      uint32_t ssrc_to_filter_max,
+      rtc::ArrayView<const RtpExtension> audio_extensions,
+      rtc::ArrayView<const RtpExtension> video_extensions);
+  LayerFilteringTransport(
+      TaskQueueBase* task_queue,
+      std::unique_ptr<SimulatedPacketReceiverInterface> pipe,
+      Call* send_call,
+      uint8_t vp8_video_payload_type,
+      uint8_t vp9_video_payload_type,
+      int selected_tl,
+      int selected_sl,
+      const std::map<uint8_t, MediaType>& payload_type_map,
+      rtc::ArrayView<const RtpExtension> audio_extensions,
+      rtc::ArrayView<const RtpExtension> video_extensions);
+  bool DiscardedLastPacket() const;
+  bool SendRtp(rtc::ArrayView<const uint8_t> data,
+               const PacketOptions& options) override;
+
+ private:
+  // Used to distinguish between VP8 and VP9.
+  const uint8_t vp8_video_payload_type_;
+  const uint8_t vp9_video_payload_type_;
+  const std::unique_ptr<VideoRtpDepacketizer> vp8_depacketizer_;
+  const std::unique_ptr<VideoRtpDepacketizer> vp9_depacketizer_;
+  // Discard or invalidate all temporal/spatial layers with id greater than the
+  // selected one. -1 to disable filtering.
+  const int selected_tl_;
+  const int selected_sl_;
+  bool discarded_last_packet_;
+  int num_active_spatial_layers_;
+  const uint32_t ssrc_to_filter_min_;
+  const uint32_t ssrc_to_filter_max_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_LAYER_FILTERING_TRANSPORT_H_