Adding upstream version 115.7.0esr.upstream/115.7.0esrupstream

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

1 files changed, 388 insertions, 0 deletions

diff --git a/third_party/libwebrtc/call/degraded_call.cc b/third_party/libwebrtc/call/degraded_call.cc
new file mode 100644
index 0000000000..99c32296c3
--- /dev/null
+++ b/third_party/libwebrtc/call/degraded_call.cc
@@ -0,0 +1,388 @@
+/*
+ *  Copyright (c) 2018 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 "call/degraded_call.h"
+
+#include <memory>
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "api/sequence_checker.h"
+#include "modules/rtp_rtcp/source/rtp_util.h"
+#include "rtc_base/event.h"
+
+namespace webrtc {
+
+DegradedCall::FakeNetworkPipeOnTaskQueue::FakeNetworkPipeOnTaskQueue(
+    TaskQueueBase* task_queue,
+    rtc::scoped_refptr<PendingTaskSafetyFlag> call_alive,
+    Clock* clock,
+    std::unique_ptr<NetworkBehaviorInterface> network_behavior)
+    : clock_(clock),
+      task_queue_(task_queue),
+      call_alive_(std::move(call_alive)),
+      pipe_(clock, std::move(network_behavior)) {}
+
+void DegradedCall::FakeNetworkPipeOnTaskQueue::SendRtp(
+    const uint8_t* packet,
+    size_t length,
+    const PacketOptions& options,
+    Transport* transport) {
+  pipe_.SendRtp(packet, length, options, transport);
+  Process();
+}
+
+void DegradedCall::FakeNetworkPipeOnTaskQueue::SendRtcp(const uint8_t* packet,
+                                                        size_t length,
+                                                        Transport* transport) {
+  pipe_.SendRtcp(packet, length, transport);
+  Process();
+}
+
+void DegradedCall::FakeNetworkPipeOnTaskQueue::AddActiveTransport(
+    Transport* transport) {
+  pipe_.AddActiveTransport(transport);
+}
+
+void DegradedCall::FakeNetworkPipeOnTaskQueue::RemoveActiveTransport(
+    Transport* transport) {
+  pipe_.RemoveActiveTransport(transport);
+}
+
+bool DegradedCall::FakeNetworkPipeOnTaskQueue::Process() {
+  pipe_.Process();
+  auto time_to_next = pipe_.TimeUntilNextProcess();
+  if (!time_to_next) {
+    // Packet was probably sent immediately.
+    return false;
+  }
+
+  task_queue_->PostTask(SafeTask(call_alive_, [this, time_to_next] {
+    RTC_DCHECK_RUN_ON(task_queue_);
+    int64_t next_process_time = *time_to_next + clock_->TimeInMilliseconds();
+    if (!next_process_ms_ || next_process_time < *next_process_ms_) {
+      next_process_ms_ = next_process_time;
+      task_queue_->PostDelayedHighPrecisionTask(
+          SafeTask(call_alive_,
+                   [this] {
+                     RTC_DCHECK_RUN_ON(task_queue_);
+                     if (!Process()) {
+                       next_process_ms_.reset();
+                     }
+                   }),
+          TimeDelta::Millis(*time_to_next));
+    }
+  }));
+
+  return true;
+}
+
+DegradedCall::FakeNetworkPipeTransportAdapter::FakeNetworkPipeTransportAdapter(
+    FakeNetworkPipeOnTaskQueue* fake_network,
+    Call* call,
+    Clock* clock,
+    Transport* real_transport)
+    : network_pipe_(fake_network),
+      call_(call),
+      clock_(clock),
+      real_transport_(real_transport) {
+  network_pipe_->AddActiveTransport(real_transport);
+}
+
+DegradedCall::FakeNetworkPipeTransportAdapter::
+    ~FakeNetworkPipeTransportAdapter() {
+  network_pipe_->RemoveActiveTransport(real_transport_);
+}
+
+bool DegradedCall::FakeNetworkPipeTransportAdapter::SendRtp(
+    const uint8_t* packet,
+    size_t length,
+    const PacketOptions& options) {
+  // A call here comes from the RTP stack (probably pacer). We intercept it and
+  // put it in the fake network pipe instead, but report to Call that is has
+  // been sent, so that the bandwidth estimator sees the delay we add.
+  network_pipe_->SendRtp(packet, length, options, real_transport_);
+  if (options.packet_id != -1) {
+    rtc::SentPacket sent_packet;
+    sent_packet.packet_id = options.packet_id;
+    sent_packet.send_time_ms = clock_->TimeInMilliseconds();
+    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;
+    call_->OnSentPacket(sent_packet);
+  }
+  return true;
+}
+
+bool DegradedCall::FakeNetworkPipeTransportAdapter::SendRtcp(
+    const uint8_t* packet,
+    size_t length) {
+  network_pipe_->SendRtcp(packet, length, real_transport_);
+  return true;
+}
+
+/* Mozilla: Avoid this since it could use GetRealTimeClock().
+DegradedCall::DegradedCall(
+    std::unique_ptr<Call> call,
+    const std::vector<TimeScopedNetworkConfig>& send_configs,
+    const std::vector<TimeScopedNetworkConfig>& receive_configs)
+    : clock_(Clock::GetRealTimeClock()),
+      call_(std::move(call)),
+      call_alive_(PendingTaskSafetyFlag::CreateDetached()),
+      send_config_index_(0),
+      send_configs_(send_configs),
+      send_simulated_network_(nullptr),
+      receive_config_index_(0),
+      receive_configs_(receive_configs) {
+  if (!receive_configs_.empty()) {
+    auto network = std::make_unique<SimulatedNetwork>(receive_configs_[0]);
+    receive_simulated_network_ = network.get();
+    receive_pipe_ =
+        std::make_unique<webrtc::FakeNetworkPipe>(clock_, std::move(network));
+    receive_pipe_->SetReceiver(call_->Receiver());
+    if (receive_configs_.size() > 1) {
+      call_->network_thread()->PostDelayedTask(
+          SafeTask(call_alive_, [this] { UpdateReceiveNetworkConfig(); }),
+          receive_configs_[0].duration);
+    }
+  }
+  if (!send_configs_.empty()) {
+    auto network = std::make_unique<SimulatedNetwork>(send_configs_[0]);
+    send_simulated_network_ = network.get();
+    send_pipe_ = std::make_unique<FakeNetworkPipeOnTaskQueue>(
+        call_->network_thread(), call_alive_, clock_, std::move(network));
+    if (send_configs_.size() > 1) {
+      call_->network_thread()->PostDelayedTask(
+          SafeTask(call_alive_, [this] { UpdateSendNetworkConfig(); }),
+          send_configs_[0].duration);
+    }
+  }
+}
+*/
+
+DegradedCall::~DegradedCall() {
+  RTC_DCHECK_RUN_ON(call_->worker_thread());
+  // Thread synchronization is required to call `SetNotAlive`.
+  // Otherwise, when the `DegradedCall` object is destroyed but
+  // `SetNotAlive` has not yet been called,
+  // another Closure guarded by `call_alive_` may be called.
+  rtc::Event event;
+  call_->network_thread()->PostTask(
+      [flag = std::move(call_alive_), &event]() mutable {
+        flag->SetNotAlive();
+        event.Set();
+      });
+  event.Wait(rtc::Event::kForever);
+}
+
+AudioSendStream* DegradedCall::CreateAudioSendStream(
+    const AudioSendStream::Config& config) {
+  if (!send_configs_.empty()) {
+    auto transport_adapter = std::make_unique<FakeNetworkPipeTransportAdapter>(
+        send_pipe_.get(), call_.get(), clock_, config.send_transport);
+    AudioSendStream::Config degrade_config = config;
+    degrade_config.send_transport = transport_adapter.get();
+    AudioSendStream* send_stream = call_->CreateAudioSendStream(degrade_config);
+    if (send_stream) {
+      audio_send_transport_adapters_[send_stream] =
+          std::move(transport_adapter);
+    }
+    return send_stream;
+  }
+  return call_->CreateAudioSendStream(config);
+}
+
+void DegradedCall::DestroyAudioSendStream(AudioSendStream* send_stream) {
+  call_->DestroyAudioSendStream(send_stream);
+  audio_send_transport_adapters_.erase(send_stream);
+}
+
+AudioReceiveStreamInterface* DegradedCall::CreateAudioReceiveStream(
+    const AudioReceiveStreamInterface::Config& config) {
+  return call_->CreateAudioReceiveStream(config);
+}
+
+void DegradedCall::DestroyAudioReceiveStream(
+    AudioReceiveStreamInterface* receive_stream) {
+  call_->DestroyAudioReceiveStream(receive_stream);
+}
+
+VideoSendStream* DegradedCall::CreateVideoSendStream(
+    VideoSendStream::Config config,
+    VideoEncoderConfig encoder_config) {
+  std::unique_ptr<FakeNetworkPipeTransportAdapter> transport_adapter;
+  if (!send_configs_.empty()) {
+    transport_adapter = std::make_unique<FakeNetworkPipeTransportAdapter>(
+        send_pipe_.get(), call_.get(), clock_, config.send_transport);
+    config.send_transport = transport_adapter.get();
+  }
+  VideoSendStream* send_stream = call_->CreateVideoSendStream(
+      std::move(config), std::move(encoder_config));
+  if (send_stream && transport_adapter) {
+    video_send_transport_adapters_[send_stream] = std::move(transport_adapter);
+  }
+  return send_stream;
+}
+
+VideoSendStream* DegradedCall::CreateVideoSendStream(
+    VideoSendStream::Config config,
+    VideoEncoderConfig encoder_config,
+    std::unique_ptr<FecController> fec_controller) {
+  std::unique_ptr<FakeNetworkPipeTransportAdapter> transport_adapter;
+  if (!send_configs_.empty()) {
+    transport_adapter = std::make_unique<FakeNetworkPipeTransportAdapter>(
+        send_pipe_.get(), call_.get(), clock_, config.send_transport);
+    config.send_transport = transport_adapter.get();
+  }
+  VideoSendStream* send_stream = call_->CreateVideoSendStream(
+      std::move(config), std::move(encoder_config), std::move(fec_controller));
+  if (send_stream && transport_adapter) {
+    video_send_transport_adapters_[send_stream] = std::move(transport_adapter);
+  }
+  return send_stream;
+}
+
+void DegradedCall::DestroyVideoSendStream(VideoSendStream* send_stream) {
+  call_->DestroyVideoSendStream(send_stream);
+  video_send_transport_adapters_.erase(send_stream);
+}
+
+VideoReceiveStreamInterface* DegradedCall::CreateVideoReceiveStream(
+    VideoReceiveStreamInterface::Config configuration) {
+  return call_->CreateVideoReceiveStream(std::move(configuration));
+}
+
+void DegradedCall::DestroyVideoReceiveStream(
+    VideoReceiveStreamInterface* receive_stream) {
+  call_->DestroyVideoReceiveStream(receive_stream);
+}
+
+FlexfecReceiveStream* DegradedCall::CreateFlexfecReceiveStream(
+    const FlexfecReceiveStream::Config config) {
+  return call_->CreateFlexfecReceiveStream(std::move(config));
+}
+
+void DegradedCall::DestroyFlexfecReceiveStream(
+    FlexfecReceiveStream* receive_stream) {
+  call_->DestroyFlexfecReceiveStream(receive_stream);
+}
+
+void DegradedCall::AddAdaptationResource(
+    rtc::scoped_refptr<Resource> resource) {
+  call_->AddAdaptationResource(std::move(resource));
+}
+
+PacketReceiver* DegradedCall::Receiver() {
+  if (!receive_configs_.empty()) {
+    return this;
+  }
+  return call_->Receiver();
+}
+
+RtpTransportControllerSendInterface*
+DegradedCall::GetTransportControllerSend() {
+  return call_->GetTransportControllerSend();
+}
+
+Call::Stats DegradedCall::GetStats() const {
+  return call_->GetStats();
+}
+
+const FieldTrialsView& DegradedCall::trials() const {
+  return call_->trials();
+}
+
+TaskQueueBase* DegradedCall::network_thread() const {
+  return call_->network_thread();
+}
+
+TaskQueueBase* DegradedCall::worker_thread() const {
+  return call_->worker_thread();
+}
+
+void DegradedCall::SignalChannelNetworkState(MediaType media,
+                                             NetworkState state) {
+  call_->SignalChannelNetworkState(media, state);
+}
+
+void DegradedCall::OnAudioTransportOverheadChanged(
+    int transport_overhead_per_packet) {
+  call_->OnAudioTransportOverheadChanged(transport_overhead_per_packet);
+}
+
+void DegradedCall::OnLocalSsrcUpdated(AudioReceiveStreamInterface& stream,
+                                      uint32_t local_ssrc) {
+  call_->OnLocalSsrcUpdated(stream, local_ssrc);
+}
+
+void DegradedCall::OnLocalSsrcUpdated(VideoReceiveStreamInterface& stream,
+                                      uint32_t local_ssrc) {
+  call_->OnLocalSsrcUpdated(stream, local_ssrc);
+}
+
+void DegradedCall::OnLocalSsrcUpdated(FlexfecReceiveStream& stream,
+                                      uint32_t local_ssrc) {
+  call_->OnLocalSsrcUpdated(stream, local_ssrc);
+}
+
+void DegradedCall::OnUpdateSyncGroup(AudioReceiveStreamInterface& stream,
+                                     absl::string_view sync_group) {
+  call_->OnUpdateSyncGroup(stream, sync_group);
+}
+
+void DegradedCall::OnSentPacket(const rtc::SentPacket& sent_packet) {
+  if (!send_configs_.empty()) {
+    // If we have a degraded send-transport, we have already notified call
+    // about the supposed network send time. Discard the actual network send
+    // time in order to properly fool the BWE.
+    return;
+  }
+  call_->OnSentPacket(sent_packet);
+}
+
+void DegradedCall::DeliverRtpPacket(
+    MediaType media_type,
+    RtpPacketReceived packet,
+    OnUndemuxablePacketHandler undemuxable_packet_handler) {
+  RTC_DCHECK_RUN_ON(&received_packet_sequence_checker_);
+  receive_pipe_->DeliverRtpPacket(media_type, std::move(packet),
+                                  std::move(undemuxable_packet_handler));
+  receive_pipe_->Process();
+}
+
+void DegradedCall::DeliverRtcpPacket(rtc::CopyOnWriteBuffer packet) {
+  RTC_DCHECK_RUN_ON(&received_packet_sequence_checker_);
+  receive_pipe_->DeliverRtcpPacket(std::move(packet));
+  receive_pipe_->Process();
+}
+
+void DegradedCall::SetClientBitratePreferences(
+    const webrtc::BitrateSettings& preferences) {
+  call_->SetClientBitratePreferences(preferences);
+}
+
+void DegradedCall::UpdateSendNetworkConfig() {
+  send_config_index_ = (send_config_index_ + 1) % send_configs_.size();
+  send_simulated_network_->SetConfig(send_configs_[send_config_index_]);
+  call_->network_thread()->PostDelayedTask(
+      SafeTask(call_alive_, [this] { UpdateSendNetworkConfig(); }),
+      send_configs_[send_config_index_].duration);
+}
+
+void DegradedCall::UpdateReceiveNetworkConfig() {
+  receive_config_index_ = (receive_config_index_ + 1) % receive_configs_.size();
+  receive_simulated_network_->SetConfig(
+      receive_configs_[receive_config_index_]);
+  call_->network_thread()->PostDelayedTask(
+      SafeTask(call_alive_, [this] { UpdateReceiveNetworkConfig(); }),
+      receive_configs_[receive_config_index_].duration);
+}
+}  // namespace webrtc