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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/pc/test/rtp_transport_test_util.h | |
parent | Initial commit. (diff) | |
download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip |
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/pc/test/rtp_transport_test_util.h')
-rw-r--r-- | third_party/libwebrtc/pc/test/rtp_transport_test_util.h | 98 |
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_ |