From 6bf0a5cb5034a7e684dcc3500e841785237ce2dd Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Sun, 7 Apr 2024 19:32:43 +0200 Subject: Adding upstream version 1:115.7.0. Signed-off-by: Daniel Baumann --- .../libwebrtc/pc/test/rtp_transport_test_util.h | 78 ++++++++++++++++++++++ 1 file changed, 78 insertions(+) create mode 100644 third_party/libwebrtc/pc/test/rtp_transport_test_util.h (limited to 'third_party/libwebrtc/pc/test/rtp_transport_test_util.h') 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..0353b74754 --- /dev/null +++ b/third_party/libwebrtc/pc/test/rtp_transport_test_util.h @@ -0,0 +1,78 @@ +/* + * 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 "call/rtp_packet_sink_interface.h" +#include "modules/rtp_rtcp/source/rtp_packet_received.h" +#include "pc/rtp_transport_internal.h" +#include "rtc_base/third_party/sigslot/sigslot.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 sigslot::has_slots<> { + public: + TransportObserver() {} + + explicit TransportObserver(RtpTransportInternal* rtp_transport) { + rtp_transport->SignalRtcpPacketReceived.connect( + this, &TransportObserver::OnRtcpPacketReceived); + rtp_transport->SignalReadyToSend.connect(this, + &TransportObserver::OnReadyToSend); + } + + // RtpPacketInterface override. + void OnRtpPacket(const RtpPacketReceived& packet) override { + rtp_count_++; + last_recv_rtp_packet_ = packet.Buffer(); + } + + 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 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) { + 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_; } + + private: + bool ready_to_send_ = false; + int 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_; +}; + +} // namespace webrtc + +#endif // PC_TEST_RTP_TRANSPORT_TEST_UTIL_H_ -- cgit v1.2.3