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/*
* 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](RtpPacketReceived& packet) { OnUndemuxableRtpPacket(packet); });
rtp_transport->SubscribeSentPacket(this,
[this](const rtc::SentPacket& packet) {
sent_packet_count_++;
if (action_on_sent_packet_) {
action_on_sent_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_; }
int sent_packet_count() const { return sent_packet_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);
}
void SetActionOnSentPacket(absl::AnyInvocable<void()> action) {
action_on_sent_packet_ = std::move(action);
}
private:
bool ready_to_send_ = false;
int rtp_count_ = 0;
int un_demuxable_rtp_count_ = 0;
int rtcp_count_ = 0;
int sent_packet_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_;
absl::AnyInvocable<void()> action_on_sent_packet_;
};
} // namespace webrtc
#endif // PC_TEST_RTP_TRANSPORT_TEST_UTIL_H_
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