a6544377bc
This test drives the new tools_webrtc/remove_extra_namespace.py tool. Bug: None Change-Id: I9b590aa1213e4cace2d64d555f4dafd893f03606 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327021 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41141}
110 lines
3.6 KiB
C++
110 lines
3.6 KiB
C++
/*
|
|
* 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_
|