Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 90 insertions, 0 deletions

diff --git a/third_party/libwebrtc/test/direct_transport.h b/third_party/libwebrtc/test/direct_transport.h
new file mode 100644
index 0000000000..1e14c1c1f4
--- /dev/null
+++ b/third_party/libwebrtc/test/direct_transport.h
@@ -0,0 +1,90 @@
+/*
+ *  Copyright (c) 2013 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_DIRECT_TRANSPORT_H_
+#define TEST_DIRECT_TRANSPORT_H_
+
+#include <memory>
+
+#include "api/call/transport.h"
+#include "api/sequence_checker.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/test/simulated_network.h"
+#include "call/call.h"
+#include "call/simulated_packet_receiver.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/task_utils/repeating_task.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+class PacketReceiver;
+
+namespace test {
+class Demuxer {
+ public:
+  explicit Demuxer(const std::map<uint8_t, MediaType>& payload_type_map);
+  ~Demuxer() = default;
+
+  Demuxer(const Demuxer&) = delete;
+  Demuxer& operator=(const Demuxer&) = delete;
+
+  MediaType GetMediaType(const uint8_t* packet_data,
+                         size_t packet_length) const;
+  const std::map<uint8_t, MediaType> payload_type_map_;
+};
+
+// Objects of this class are expected to be allocated and destroyed  on the
+// same task-queue - the one that's passed in via the constructor.
+class DirectTransport : public Transport {
+ public:
+  DirectTransport(TaskQueueBase* task_queue,
+                  std::unique_ptr<SimulatedPacketReceiverInterface> pipe,
+                  Call* send_call,
+                  const std::map<uint8_t, MediaType>& payload_type_map,
+                  rtc::ArrayView<const RtpExtension> audio_extensions,
+                  rtc::ArrayView<const RtpExtension> video_extensions);
+
+  ~DirectTransport() override;
+
+  // TODO(holmer): Look into moving this to the constructor.
+  virtual void SetReceiver(PacketReceiver* receiver);
+
+  // Backwards compatibility using statements.
+  // TODO(https://bugs.webrtc.org/15410): Remove when not needed.
+  using Transport::SendRtcp;
+  using Transport::SendRtp;
+
+  bool SendRtp(rtc::ArrayView<const uint8_t> data,
+               const PacketOptions& options) override;
+  bool SendRtcp(rtc::ArrayView<const uint8_t> data) override;
+
+  int GetAverageDelayMs();
+
+ private:
+  void ProcessPackets() RTC_EXCLUSIVE_LOCKS_REQUIRED(&process_lock_);
+  void LegacySendPacket(const uint8_t* data, size_t length);
+  void Start();
+
+  Call* const send_call_;
+
+  TaskQueueBase* const task_queue_;
+
+  Mutex process_lock_;
+  RepeatingTaskHandle next_process_task_ RTC_GUARDED_BY(&process_lock_);
+
+  const Demuxer demuxer_;
+  const std::unique_ptr<SimulatedPacketReceiverInterface> fake_network_;
+  const RtpHeaderExtensionMap audio_extensions_;
+  const RtpHeaderExtensionMap video_extensions_;
+};
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_DIRECT_TRANSPORT_H_