Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 145 insertions, 0 deletions

diff --git a/third_party/libwebrtc/test/direct_transport.cc b/third_party/libwebrtc/test/direct_transport.cc
new file mode 100644
index 0000000000..260497947c
--- /dev/null
+++ b/third_party/libwebrtc/test/direct_transport.cc
@@ -0,0 +1,145 @@
+/*
+ *  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