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diff --git a/third_party/libwebrtc/test/direct_transport.cc b/third_party/libwebrtc/test/direct_transport.cc
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+/*
+ * Copyright (c) 2013 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.
+ */
+#include "test/direct_transport.h"
+
+#include "api/media_types.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/units/time_delta.h"
+#include "call/call.h"
+#include "call/fake_network_pipe.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtp_util.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/task_utils/repeating_task.h"
+#include "rtc_base/time_utils.h"
+
+namespace webrtc {
+namespace test {
+
+Demuxer::Demuxer(const std::map<uint8_t, MediaType>& payload_type_map)
+ : payload_type_map_(payload_type_map) {}
+
+MediaType Demuxer::GetMediaType(const uint8_t* packet_data,
+ const size_t packet_length) const {
+ if (IsRtpPacket(rtc::MakeArrayView(packet_data, packet_length))) {
+ RTC_CHECK_GE(packet_length, 2);
+ const uint8_t payload_type = packet_data[1] & 0x7f;
+ std::map<uint8_t, MediaType>::const_iterator it =
+ payload_type_map_.find(payload_type);
+ RTC_CHECK(it != payload_type_map_.end())
+ << "payload type " << static_cast<int>(payload_type) << " unknown.";
+ return it->second;
+ }
+ return MediaType::ANY;
+}
+
+DirectTransport::DirectTransport(
+ TaskQueueBase* task_queue,
+ std::unique_ptr<SimulatedPacketReceiverInterface> pipe,
+ Call* send_call,
+ const std::map<uint8_t, MediaType>& payload_type_map,
+ rtc::ArrayView<const RtpExtension> audio_extensions,
+ rtc::ArrayView<const RtpExtension> video_extensions)
+ : send_call_(send_call),
+ task_queue_(task_queue),
+ demuxer_(payload_type_map),
+ fake_network_(std::move(pipe)),
+ audio_extensions_(audio_extensions),
+ video_extensions_(video_extensions) {
+ Start();
+}
+
+DirectTransport::~DirectTransport() {
+ next_process_task_.Stop();
+}
+
+void DirectTransport::SetReceiver(PacketReceiver* receiver) {
+ fake_network_->SetReceiver(receiver);
+}
+
+bool DirectTransport::SendRtp(const uint8_t* data,
+ size_t length,
+ const PacketOptions& options) {
+ if (send_call_) {
+ rtc::SentPacket sent_packet(options.packet_id, rtc::TimeMillis());
+ sent_packet.info.included_in_feedback = options.included_in_feedback;
+ sent_packet.info.included_in_allocation = options.included_in_allocation;
+ sent_packet.info.packet_size_bytes = length;
+ sent_packet.info.packet_type = rtc::PacketType::kData;
+ send_call_->OnSentPacket(sent_packet);
+ }
+
+ const RtpHeaderExtensionMap* extensions = nullptr;
+ MediaType media_type = demuxer_.GetMediaType(data, length);
+ switch (demuxer_.GetMediaType(data, length)) {
+ case webrtc::MediaType::AUDIO:
+ extensions = &audio_extensions_;
+ break;
+ case webrtc::MediaType::VIDEO:
+ extensions = &video_extensions_;
+ break;
+ default:
+ RTC_CHECK_NOTREACHED();
+ }
+ RtpPacketReceived packet(extensions, Timestamp::Micros(rtc::TimeMicros()));
+ if (media_type == MediaType::VIDEO) {
+ packet.set_payload_type_frequency(kVideoPayloadTypeFrequency);
+ }
+ RTC_CHECK(packet.Parse(rtc::CopyOnWriteBuffer(data, length)));
+ fake_network_->DeliverRtpPacket(
+ media_type, std::move(packet),
+ [](const RtpPacketReceived& packet) { return false; });
+
+ MutexLock lock(&process_lock_);
+ if (!next_process_task_.Running())
+ ProcessPackets();
+ return true;
+}
+
+bool DirectTransport::SendRtcp(const uint8_t* data, size_t length) {
+ fake_network_->DeliverRtcpPacket(rtc::CopyOnWriteBuffer(data, length));
+ MutexLock lock(&process_lock_);
+ if (!next_process_task_.Running())
+ ProcessPackets();
+ return true;
+}
+
+int DirectTransport::GetAverageDelayMs() {
+ return fake_network_->AverageDelay();
+}
+
+void DirectTransport::Start() {
+ RTC_DCHECK(task_queue_);
+ if (send_call_) {
+ send_call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkUp);
+ send_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
+ }
+}
+
+void DirectTransport::ProcessPackets() {
+ absl::optional<int64_t> initial_delay_ms =
+ fake_network_->TimeUntilNextProcess();
+ if (initial_delay_ms == absl::nullopt)
+ return;
+
+ next_process_task_ = RepeatingTaskHandle::DelayedStart(
+ task_queue_, TimeDelta::Millis(*initial_delay_ms), [this] {
+ fake_network_->Process();
+ if (auto delay_ms = fake_network_->TimeUntilNextProcess())
+ return TimeDelta::Millis(*delay_ms);
+ // Otherwise stop the task.
+ MutexLock lock(&process_lock_);
+ next_process_task_.Stop();
+ // Since this task is stopped, return value doesn't matter.
+ return TimeDelta::Zero();
+ });
+}
+} // namespace test
+} // namespace webrtc