blob: 69f993eb4b052ca05cc6f5b707c0841981fd02ab [file] [log] [blame]
/*
* 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.
*/
// This is EXPERIMENTAL interface for media transport.
//
// The goal is to refactor WebRTC code so that audio and video frames
// are sent / received through the media transport interface. This will
// enable different media transport implementations, including QUIC-based
// media transport.
#include "api/media_transport_interface.h"
#include <cstdint>
#include <utility>
#include "api/datagram_transport_interface.h"
namespace webrtc {
MediaTransportSettings::MediaTransportSettings() = default;
MediaTransportSettings::MediaTransportSettings(const MediaTransportSettings&) =
default;
MediaTransportSettings& MediaTransportSettings::operator=(
const MediaTransportSettings&) = default;
MediaTransportSettings::~MediaTransportSettings() = default;
SendDataParams::SendDataParams() = default;
SendDataParams::SendDataParams(const SendDataParams&) = default;
RTCErrorOr<std::unique_ptr<MediaTransportInterface>>
MediaTransportFactory::CreateMediaTransport(
rtc::PacketTransportInternal* packet_transport,
rtc::Thread* network_thread,
const MediaTransportSettings& settings) {
return std::unique_ptr<MediaTransportInterface>(nullptr);
}
RTCErrorOr<std::unique_ptr<MediaTransportInterface>>
MediaTransportFactory::CreateMediaTransport(
rtc::Thread* network_thread,
const MediaTransportSettings& settings) {
return std::unique_ptr<MediaTransportInterface>(nullptr);
}
RTCErrorOr<std::unique_ptr<DatagramTransportInterface>>
MediaTransportFactory::CreateDatagramTransport(
rtc::Thread* network_thread,
const MediaTransportSettings& settings) {
return std::unique_ptr<DatagramTransportInterface>(nullptr);
}
std::string MediaTransportFactory::GetTransportName() const {
return "";
}
MediaTransportInterface::MediaTransportInterface() = default;
MediaTransportInterface::~MediaTransportInterface() = default;
absl::optional<std::string>
MediaTransportInterface::GetTransportParametersOffer() const {
return absl::nullopt;
}
void MediaTransportInterface::Connect(
rtc::PacketTransportInternal* packet_transport) {}
void MediaTransportInterface::SetKeyFrameRequestCallback(
MediaTransportKeyFrameRequestCallback* callback) {}
absl::optional<TargetTransferRate>
MediaTransportInterface::GetLatestTargetTransferRate() {
return absl::nullopt;
}
void MediaTransportInterface::AddNetworkChangeCallback(
MediaTransportNetworkChangeCallback* callback) {}
void MediaTransportInterface::RemoveNetworkChangeCallback(
MediaTransportNetworkChangeCallback* callback) {}
void MediaTransportInterface::SetFirstAudioPacketReceivedObserver(
AudioPacketReceivedObserver* observer) {}
void MediaTransportInterface::AddTargetTransferRateObserver(
TargetTransferRateObserver* observer) {}
void MediaTransportInterface::RemoveTargetTransferRateObserver(
TargetTransferRateObserver* observer) {}
void MediaTransportInterface::AddRttObserver(
MediaTransportRttObserver* observer) {}
void MediaTransportInterface::RemoveRttObserver(
MediaTransportRttObserver* observer) {}
size_t MediaTransportInterface::GetAudioPacketOverhead() const {
return 0;
}
void MediaTransportInterface::SetAllocatedBitrateLimits(
const MediaTransportAllocatedBitrateLimits& limits) {}
} // namespace webrtc