Adding upstream version 124.0.1.upstream/124.0.1

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

1 files changed, 98 insertions, 0 deletions

diff --git a/third_party/libwebrtc/pc/test/rtp_transport_test_util.h b/third_party/libwebrtc/pc/test/rtp_transport_test_util.h
new file mode 100644
index 0000000000..593ee002c9
--- /dev/null
+++ b/third_party/libwebrtc/pc/test/rtp_transport_test_util.h
@@ -0,0 +1,98 @@
+/*
+ *  Copyright 2017 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 PC_TEST_RTP_TRANSPORT_TEST_UTIL_H_
+#define PC_TEST_RTP_TRANSPORT_TEST_UTIL_H_
+
+#include <utility>
+
+#include "call/rtp_packet_sink_interface.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "pc/rtp_transport_internal.h"
+
+namespace webrtc {
+
+// Used to handle the signals when the RtpTransport receives an RTP/RTCP packet.
+// Used in Rtp/Srtp/DtlsTransport unit tests.
+class TransportObserver : public RtpPacketSinkInterface {
+ public:
+  TransportObserver() {}
+
+  explicit TransportObserver(RtpTransportInternal* rtp_transport) {
+    rtp_transport->SubscribeRtcpPacketReceived(
+        this, [this](rtc::CopyOnWriteBuffer* buffer, int64_t packet_time_ms) {
+          OnRtcpPacketReceived(buffer, packet_time_ms);
+        });
+    rtp_transport->SubscribeReadyToSend(
+        this, [this](bool arg) { OnReadyToSend(arg); });
+    rtp_transport->SetUnDemuxableRtpPacketReceivedHandler(
+        [this](webrtc::RtpPacketReceived& packet) {
+          OnUndemuxableRtpPacket(packet);
+        });
+  }
+
+  // RtpPacketInterface override.
+  void OnRtpPacket(const RtpPacketReceived& packet) override {
+    rtp_count_++;
+    last_recv_rtp_packet_ = packet.Buffer();
+  }
+
+  void OnUndemuxableRtpPacket(const RtpPacketReceived& packet) {
+    un_demuxable_rtp_count_++;
+  }
+
+  void OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
+                            int64_t packet_time_us) {
+    rtcp_count_++;
+    last_recv_rtcp_packet_ = *packet;
+  }
+
+  int rtp_count() const { return rtp_count_; }
+  int un_demuxable_rtp_count() const { return un_demuxable_rtp_count_; }
+  int rtcp_count() const { return rtcp_count_; }
+
+  rtc::CopyOnWriteBuffer last_recv_rtp_packet() {
+    return last_recv_rtp_packet_;
+  }
+
+  rtc::CopyOnWriteBuffer last_recv_rtcp_packet() {
+    return last_recv_rtcp_packet_;
+  }
+
+  void OnReadyToSend(bool ready) {
+    if (action_on_ready_to_send_) {
+      action_on_ready_to_send_(ready);
+    }
+    ready_to_send_signal_count_++;
+    ready_to_send_ = ready;
+  }
+
+  bool ready_to_send() { return ready_to_send_; }
+
+  int ready_to_send_signal_count() { return ready_to_send_signal_count_; }
+
+  void SetActionOnReadyToSend(absl::AnyInvocable<void(bool)> action) {
+    action_on_ready_to_send_ = std::move(action);
+  }
+
+ private:
+  bool ready_to_send_ = false;
+  int rtp_count_ = 0;
+  int un_demuxable_rtp_count_ = 0;
+  int rtcp_count_ = 0;
+  int ready_to_send_signal_count_ = 0;
+  rtc::CopyOnWriteBuffer last_recv_rtp_packet_;
+  rtc::CopyOnWriteBuffer last_recv_rtcp_packet_;
+  absl::AnyInvocable<void(bool)> action_on_ready_to_send_;
+};
+
+}  // namespace webrtc
+
+#endif  // PC_TEST_RTP_TRANSPORT_TEST_UTIL_H_