1 files changed, 241 insertions, 0 deletions
diff --git a/third_party/libwebrtc/test/scenario/audio_stream.cc b/third_party/libwebrtc/test/scenario/audio_stream.cc
new file mode 100644
index 0000000000..5f2eff12ff
--- /dev/null
+++ b/third_party/libwebrtc/test/scenario/audio_stream.cc
@@ -0,0 +1,241 @@
+/*
+ *  Copyright 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 "test/scenario/audio_stream.h"
+
+#include "absl/memory/memory.h"
+#include "test/call_test.h"
+
+#if WEBRTC_ENABLE_PROTOBUF
+RTC_PUSH_IGNORING_WUNDEF()
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/modules/audio_coding/audio_network_adaptor/config.pb.h"
+#else
+#include "modules/audio_coding/audio_network_adaptor/config.pb.h"
+#endif
+RTC_POP_IGNORING_WUNDEF()
+#endif
+
+namespace webrtc {
+namespace test {
+namespace {
+enum : int {  // The first valid value is 1.
+  kTransportSequenceNumberExtensionId = 1,
+  kAbsSendTimeExtensionId
+};
+
+absl::optional<std::string> CreateAdaptationString(
+    AudioStreamConfig::NetworkAdaptation config) {
+#if WEBRTC_ENABLE_PROTOBUF
+
+  audio_network_adaptor::config::ControllerManager cont_conf;
+  if (config.frame.max_rate_for_60_ms.IsFinite()) {
+    auto controller =
+        cont_conf.add_controllers()->mutable_frame_length_controller();
+    controller->set_fl_decreasing_packet_loss_fraction(
+        config.frame.min_packet_loss_for_decrease);
+    controller->set_fl_increasing_packet_loss_fraction(
+        config.frame.max_packet_loss_for_increase);
+
+    controller->set_fl_20ms_to_60ms_bandwidth_bps(
+        config.frame.min_rate_for_20_ms.bps<int32_t>());
+    controller->set_fl_60ms_to_20ms_bandwidth_bps(
+        config.frame.max_rate_for_60_ms.bps<int32_t>());
+
+    if (config.frame.max_rate_for_120_ms.IsFinite()) {
+      controller->set_fl_60ms_to_120ms_bandwidth_bps(
+          config.frame.min_rate_for_60_ms.bps<int32_t>());
+      controller->set_fl_120ms_to_60ms_bandwidth_bps(
+          config.frame.max_rate_for_120_ms.bps<int32_t>());
+    }
+  }
+  cont_conf.add_controllers()->mutable_bitrate_controller();
+  std::string config_string = cont_conf.SerializeAsString();
+  return config_string;
+#else
+  RTC_LOG(LS_ERROR) << "audio_network_adaptation is enabled"
+                       " but WEBRTC_ENABLE_PROTOBUF is false.\n"
+                       "Ignoring settings.";
+  return absl::nullopt;
+#endif  // WEBRTC_ENABLE_PROTOBUF
+}
+}  // namespace
+
+std::vector<RtpExtension> GetAudioRtpExtensions(
+    const AudioStreamConfig& config) {
+  std::vector<RtpExtension> extensions;
+  if (config.stream.in_bandwidth_estimation) {
+    extensions.push_back({RtpExtension::kTransportSequenceNumberUri,
+                          kTransportSequenceNumberExtensionId});
+  }
+  if (config.stream.abs_send_time) {
+    extensions.push_back(
+        {RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId});
+  }
+  return extensions;
+}
+
+SendAudioStream::SendAudioStream(
+    CallClient* sender,
+    AudioStreamConfig config,
+    rtc::scoped_refptr<AudioEncoderFactory> encoder_factory,
+    Transport* send_transport)
+    : sender_(sender), config_(config) {
+  AudioSendStream::Config send_config(send_transport);
+  ssrc_ = sender->GetNextAudioSsrc();
+  send_config.rtp.ssrc = ssrc_;
+  SdpAudioFormat::Parameters sdp_params;
+  if (config.source.channels == 2)
+    sdp_params["stereo"] = "1";
+  if (config.encoder.initial_frame_length != TimeDelta::Millis(20))
+    sdp_params["ptime"] =
+        std::to_string(config.encoder.initial_frame_length.ms());
+  if (config.encoder.enable_dtx)
+    sdp_params["usedtx"] = "1";
+
+  // SdpAudioFormat::num_channels indicates that the encoder is capable of
+  // stereo, but the actual channel count used is based on the "stereo"
+  // parameter.
+  send_config.send_codec_spec = AudioSendStream::Config::SendCodecSpec(
+      CallTest::kAudioSendPayloadType, {"opus", 48000, 2, sdp_params});
+  RTC_DCHECK_LE(config.source.channels, 2);
+  send_config.encoder_factory = encoder_factory;
+
+  bool use_fixed_rate = !config.encoder.min_rate && !config.encoder.max_rate;
+  if (use_fixed_rate)
+    send_config.send_codec_spec->target_bitrate_bps =
+        config.encoder.fixed_rate.bps();
+  if (!config.adapt.binary_proto.empty()) {
+    send_config.audio_network_adaptor_config = config.adapt.binary_proto;
+  } else if (config.network_adaptation) {
+    send_config.audio_network_adaptor_config =
+        CreateAdaptationString(config.adapt);
+  }
+  if (config.encoder.allocate_bitrate ||
+      config.stream.in_bandwidth_estimation) {
+    DataRate min_rate = DataRate::Infinity();
+    DataRate max_rate = DataRate::Infinity();
+    if (use_fixed_rate) {
+      min_rate = config.encoder.fixed_rate;
+      max_rate = config.encoder.fixed_rate;
+    } else {
+      min_rate = *config.encoder.min_rate;
+      max_rate = *config.encoder.max_rate;
+    }
+    send_config.min_bitrate_bps = min_rate.bps();
+    send_config.max_bitrate_bps = max_rate.bps();
+  }
+
+  if (config.stream.in_bandwidth_estimation) {
+    send_config.send_codec_spec->transport_cc_enabled = true;
+  }
+  send_config.rtp.extensions = GetAudioRtpExtensions(config);
+
+  sender_->SendTask([&] {
+    send_stream_ = sender_->call_->CreateAudioSendStream(send_config);
+    sender->call_->OnAudioTransportOverheadChanged(
+        sender_->transport_->packet_overhead().bytes());
+  });
+}
+
+SendAudioStream::~SendAudioStream() {
+  sender_->SendTask(
+      [this] { sender_->call_->DestroyAudioSendStream(send_stream_); });
+}
+
+void SendAudioStream::Start() {
+  sender_->SendTask([this] {
+    send_stream_->Start();
+    sender_->call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkUp);
+  });
+}
+
+void SendAudioStream::Stop() {
+  sender_->SendTask([this] { send_stream_->Stop(); });
+}
+
+void SendAudioStream::SetMuted(bool mute) {
+  sender_->SendTask([this, mute] { send_stream_->SetMuted(mute); });
+}
+
+ColumnPrinter SendAudioStream::StatsPrinter() {
+  return ColumnPrinter::Lambda(
+      "audio_target_rate",
+      [this](rtc::SimpleStringBuilder& sb) {
+        sender_->SendTask([this, &sb] {
+          AudioSendStream::Stats stats = send_stream_->GetStats();
+          sb.AppendFormat("%.0lf", stats.target_bitrate_bps / 8.0);
+        });
+      },
+      64);
+}
+
+ReceiveAudioStream::ReceiveAudioStream(
+    CallClient* receiver,
+    AudioStreamConfig config,
+    SendAudioStream* send_stream,
+    rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
+    Transport* feedback_transport)
+    : receiver_(receiver), config_(config) {
+  AudioReceiveStreamInterface::Config recv_config;
+  recv_config.rtp.local_ssrc = receiver_->GetNextAudioLocalSsrc();
+  recv_config.rtcp_send_transport = feedback_transport;
+  recv_config.rtp.remote_ssrc = send_stream->ssrc_;
+  receiver->ssrc_media_types_[recv_config.rtp.remote_ssrc] = MediaType::AUDIO;
+  recv_config.rtp.extensions = GetAudioRtpExtensions(config);
+  recv_config.decoder_factory = decoder_factory;
+  recv_config.decoder_map = {
+      {CallTest::kAudioSendPayloadType, {"opus", 48000, 2}}};
+  recv_config.sync_group = config.render.sync_group;
+  receiver_->SendTask([&] {
+    receive_stream_ = receiver_->call_->CreateAudioReceiveStream(recv_config);
+  });
+}
+ReceiveAudioStream::~ReceiveAudioStream() {
+  receiver_->SendTask(
+      [&] { receiver_->call_->DestroyAudioReceiveStream(receive_stream_); });
+}
+
+void ReceiveAudioStream::Start() {
+  receiver_->SendTask([&] {
+    receive_stream_->Start();
+    receiver_->call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkUp);
+  });
+}
+
+void ReceiveAudioStream::Stop() {
+  receiver_->SendTask([&] { receive_stream_->Stop(); });
+}
+
+AudioReceiveStreamInterface::Stats ReceiveAudioStream::GetStats() const {
+  AudioReceiveStreamInterface::Stats result;
+  receiver_->SendTask([&] {
+    result = receive_stream_->GetStats(/*get_and_clear_legacy_stats=*/true);
+  });
+  return result;
+}
+
+AudioStreamPair::~AudioStreamPair() = default;
+
+AudioStreamPair::AudioStreamPair(
+    CallClient* sender,
+    rtc::scoped_refptr<AudioEncoderFactory> encoder_factory,
+    CallClient* receiver,
+    rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
+    AudioStreamConfig config)
+    : config_(config),
+      send_stream_(sender, config, encoder_factory, sender->transport_.get()),
+      receive_stream_(receiver,
+                      config,
+                      &send_stream_,
+                      decoder_factory,
+                      receiver->transport_.get()) {}
+
+}  // namespace test
+}  // namespace webrtc