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9ded485caa
Bug: webrtc:9719 Change-Id: I9ec89fca7d4555f31b5192980f193b58d99e3b71 Reviewed-on: https://webrtc-review.googlesource.com/c/125100 Reviewed-by: Peter Slatala <psla@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26910}
100 lines
3.3 KiB
C++
100 lines
3.3 KiB
C++
/*
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* Copyright 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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// This is EXPERIMENTAL interface for media transport.
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//
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// The goal is to refactor WebRTC code so that audio and video frames
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// are sent / received through the media transport interface. This will
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// enable different media transport implementations, including QUIC-based
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// media transport.
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#include "api/media_transport_interface.h"
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#include <cstdint>
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#include <utility>
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namespace webrtc {
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MediaTransportSettings::MediaTransportSettings() = default;
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MediaTransportSettings::MediaTransportSettings(const MediaTransportSettings&) =
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default;
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MediaTransportSettings& MediaTransportSettings::operator=(
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const MediaTransportSettings&) = default;
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MediaTransportSettings::~MediaTransportSettings() = default;
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SendDataParams::SendDataParams() = default;
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SendDataParams::SendDataParams(const SendDataParams&) = default;
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RTCErrorOr<std::unique_ptr<MediaTransportInterface>>
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MediaTransportFactory::CreateMediaTransport(
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rtc::PacketTransportInternal* packet_transport,
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rtc::Thread* network_thread,
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const MediaTransportSettings& settings) {
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return std::unique_ptr<MediaTransportInterface>(nullptr);
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}
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RTCErrorOr<std::unique_ptr<MediaTransportInterface>>
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MediaTransportFactory::CreateMediaTransport(
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rtc::Thread* network_thread,
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const MediaTransportSettings& settings) {
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return std::unique_ptr<MediaTransportInterface>(nullptr);
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}
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std::string MediaTransportFactory::GetTransportName() const {
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return "";
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}
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MediaTransportInterface::MediaTransportInterface() = default;
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MediaTransportInterface::~MediaTransportInterface() = default;
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absl::optional<std::string>
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MediaTransportInterface::GetTransportParametersOffer() const {
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return absl::nullopt;
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}
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void MediaTransportInterface::Connect(
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rtc::PacketTransportInternal* packet_transport) {}
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void MediaTransportInterface::SetKeyFrameRequestCallback(
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MediaTransportKeyFrameRequestCallback* callback) {}
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absl::optional<TargetTransferRate>
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MediaTransportInterface::GetLatestTargetTransferRate() {
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return absl::nullopt;
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}
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void MediaTransportInterface::AddNetworkChangeCallback(
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MediaTransportNetworkChangeCallback* callback) {}
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void MediaTransportInterface::RemoveNetworkChangeCallback(
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MediaTransportNetworkChangeCallback* callback) {}
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void MediaTransportInterface::SetFirstAudioPacketReceivedObserver(
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AudioPacketReceivedObserver* observer) {}
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void MediaTransportInterface::AddTargetTransferRateObserver(
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TargetTransferRateObserver* observer) {}
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void MediaTransportInterface::RemoveTargetTransferRateObserver(
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TargetTransferRateObserver* observer) {}
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void MediaTransportInterface::AddRttObserver(
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MediaTransportRttObserver* observer) {}
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void MediaTransportInterface::RemoveRttObserver(
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MediaTransportRttObserver* observer) {}
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size_t MediaTransportInterface::GetAudioPacketOverhead() const {
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return 0;
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}
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void MediaTransportInterface::SetAllocatedBitrateLimits(
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const MediaTransportAllocatedBitrateLimits& limits) {}
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} // namespace webrtc
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