Adding upstream version 124.0.1.upstream/124.0.1

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

1 files changed, 99 insertions, 0 deletions

diff --git a/third_party/libwebrtc/api/voip/voip_engine.h b/third_party/libwebrtc/api/voip/voip_engine.h
new file mode 100644
index 0000000000..d223f6ad6c
--- /dev/null
+++ b/third_party/libwebrtc/api/voip/voip_engine.h
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+/*
+ *  Copyright (c) 2020 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 API_VOIP_VOIP_ENGINE_H_
+#define API_VOIP_VOIP_ENGINE_H_
+
+namespace webrtc {
+
+class VoipBase;
+class VoipCodec;
+class VoipNetwork;
+class VoipDtmf;
+class VoipStatistics;
+class VoipVolumeControl;
+
+// VoipEngine is the main interface serving as the entry point for all VoIP
+// APIs. A single instance of VoipEngine should suffice the most of the need for
+// typical VoIP applications as it handles multiple media sessions including a
+// specialized session type like ad-hoc conference. Below example code
+// describes the typical sequence of API usage. Each API header contains more
+// description on what the methods are used for.
+//
+//   // Caller is responsible of setting desired audio components.
+//   VoipEngineConfig config;
+//   config.encoder_factory = CreateBuiltinAudioEncoderFactory();
+//   config.decoder_factory = CreateBuiltinAudioDecoderFactory();
+//   config.task_queue_factory = CreateDefaultTaskQueueFactory();
+//   config.audio_device =
+//       AudioDeviceModule::Create(AudioDeviceModule::kPlatformDefaultAudio,
+//                                 config.task_queue_factory.get());
+//   config.audio_processing = AudioProcessingBuilder().Create();
+//
+//   auto voip_engine = CreateVoipEngine(std::move(config));
+//
+//   auto& voip_base = voip_engine->Base();
+//   auto& voip_codec = voip_engine->Codec();
+//   auto& voip_network = voip_engine->Network();
+//
+//   ChannelId channel = voip_base.CreateChannel(&app_transport_);
+//
+//   // After SDP offer/answer, set payload type and codecs that have been
+//   // decided through SDP negotiation.
+//   // VoipResult handling omitted here.
+//   voip_codec.SetSendCodec(channel, ...);
+//   voip_codec.SetReceiveCodecs(channel, ...);
+//
+//   // Start sending and playing RTP on voip channel.
+//   // VoipResult handling omitted here.
+//   voip_base.StartSend(channel);
+//   voip_base.StartPlayout(channel);
+//
+//   // Inject received RTP/RTCP through VoipNetwork interface.
+//   // VoipResult handling omitted here.
+//   voip_network.ReceivedRTPPacket(channel, ...);
+//   voip_network.ReceivedRTCPPacket(channel, ...);
+//
+//   // Stop and release voip channel.
+//   // VoipResult handling omitted here.
+//   voip_base.StopSend(channel);
+//   voip_base.StopPlayout(channel);
+//   voip_base.ReleaseChannel(channel);
+//
+class VoipEngine {
+ public:
+  virtual ~VoipEngine() = default;
+
+  // VoipBase is the audio session management interface that
+  // creates/releases/starts/stops an one-to-one audio media session.
+  virtual VoipBase& Base() = 0;
+
+  // VoipNetwork provides injection APIs that would enable application
+  // to send and receive RTP/RTCP packets. There is no default network module
+  // that provides RTP transmission and reception.
+  virtual VoipNetwork& Network() = 0;
+
+  // VoipCodec provides codec configuration APIs for encoder and decoders.
+  virtual VoipCodec& Codec() = 0;
+
+  // VoipDtmf provides DTMF event APIs to register and send DTMF events.
+  virtual VoipDtmf& Dtmf() = 0;
+
+  // VoipStatistics provides performance metrics around audio decoding module
+  // and jitter buffer (NetEq).
+  virtual VoipStatistics& Statistics() = 0;
+
+  // VoipVolumeControl provides various input/output volume control.
+  virtual VoipVolumeControl& VolumeControl() = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // API_VOIP_VOIP_ENGINE_H_