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ec3b9ffdb0
Bug: webrtc:9719 Change-Id: I8bef979e4073d51be7cb93d38ee0e2ae22baef0e Reviewed-on: https://webrtc-review.googlesource.com/c/121942 Reviewed-by: Peter Slatala <psla@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26594}
123 lines
4.2 KiB
C++
123 lines
4.2 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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MediaTransportEncodedVideoFrame::MediaTransportEncodedVideoFrame() = default;
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MediaTransportEncodedVideoFrame::~MediaTransportEncodedVideoFrame() = default;
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MediaTransportEncodedVideoFrame::MediaTransportEncodedVideoFrame(
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int64_t frame_id,
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std::vector<int64_t> referenced_frame_ids,
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int payload_type,
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const webrtc::EncodedImage& encoded_image)
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: payload_type_(payload_type),
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encoded_image_(encoded_image),
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frame_id_(frame_id),
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referenced_frame_ids_(std::move(referenced_frame_ids)) {}
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MediaTransportEncodedVideoFrame& MediaTransportEncodedVideoFrame::operator=(
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const MediaTransportEncodedVideoFrame&) = default;
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MediaTransportEncodedVideoFrame& MediaTransportEncodedVideoFrame::operator=(
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MediaTransportEncodedVideoFrame&&) = default;
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MediaTransportEncodedVideoFrame::MediaTransportEncodedVideoFrame(
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const MediaTransportEncodedVideoFrame& o)
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: MediaTransportEncodedVideoFrame() {
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*this = o;
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}
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MediaTransportEncodedVideoFrame::MediaTransportEncodedVideoFrame(
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MediaTransportEncodedVideoFrame&& o)
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: MediaTransportEncodedVideoFrame() {
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*this = std::move(o);
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}
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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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bool is_caller) {
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MediaTransportSettings settings;
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settings.is_caller = is_caller;
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return CreateMediaTransport(packet_transport, network_thread, settings);
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}
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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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MediaTransportInterface::MediaTransportInterface() = default;
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MediaTransportInterface::~MediaTransportInterface() = default;
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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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